diff options
Diffstat (limited to 'sys/oss')
| -rw-r--r-- | sys/oss/Makefile.am | 28 | ||||
| -rw-r--r-- | sys/oss/Makefile.in | 884 | ||||
| -rw-r--r-- | sys/oss/common.h | 43 | ||||
| -rw-r--r-- | sys/oss/gstossaudio.c | 61 | ||||
| -rw-r--r-- | sys/oss/gstossdmabuffer.h | 77 | ||||
| -rw-r--r-- | sys/oss/gstosshelper.c | 407 | ||||
| -rw-r--r-- | sys/oss/gstosshelper.h | 43 | ||||
| -rw-r--r-- | sys/oss/gstossmixer.c | 378 | ||||
| -rw-r--r-- | sys/oss/gstossmixer.h | 171 | ||||
| -rw-r--r-- | sys/oss/gstossmixerelement.c | 218 | ||||
| -rw-r--r-- | sys/oss/gstossmixerelement.h | 59 | ||||
| -rw-r--r-- | sys/oss/gstossmixertrack.c | 172 | ||||
| -rw-r--r-- | sys/oss/gstossmixertrack.h | 62 | ||||
| -rw-r--r-- | sys/oss/gstosssink.c | 565 | ||||
| -rw-r--r-- | sys/oss/gstosssink.h | 62 | ||||
| -rw-r--r-- | sys/oss/gstosssrc.c | 537 | ||||
| -rw-r--r-- | sys/oss/gstosssrc.h | 67 |
17 files changed, 3834 insertions, 0 deletions
diff --git a/sys/oss/Makefile.am b/sys/oss/Makefile.am new file mode 100644 index 0000000..c42dfea --- /dev/null +++ b/sys/oss/Makefile.am @@ -0,0 +1,28 @@ +plugin_LTLIBRARIES = libgstossaudio.la + +libgstossaudio_la_SOURCES = gstossaudio.c \ + gstosshelper.c \ + gstossmixer.c \ + gstossmixerelement.c \ + gstossmixertrack.c \ + gstosssink.c \ + gstosssrc.c + +libgstossaudio_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) +libgstossaudio_la_LIBADD = \ + $(GST_PLUGINS_BASE_LIBS) \ + -lgstinterfaces-$(GST_MAJORMINOR) \ + -lgstaudio-$(GST_MAJORMINOR) \ + $(GST_BASE_LIBS) \ + $(GST_LIBS) +libgstossaudio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) +libgstossaudio_la_LIBTOOLFLAGS = --tag=disable-static + +noinst_HEADERS = common.h \ + gstosssink.h \ + gstosssrc.h \ + gstosshelper.h \ + gstossdmabuffer.h \ + gstossmixer.h \ + gstossmixerelement.h \ + gstossmixertrack.h diff --git a/sys/oss/Makefile.in b/sys/oss/Makefile.in new file mode 100644 index 0000000..1b1a00e --- /dev/null +++ b/sys/oss/Makefile.in @@ -0,0 +1,884 @@ +# Makefile.in generated by automake 1.11.1 from Makefile.am. +# @configure_input@ + +# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002, +# 2003, 2004, 2005, 2006, 2007, 2008, 2009 Free Software Foundation, +# Inc. +# This Makefile.in is free software; the Free Software Foundation +# gives unlimited permission to copy and/or distribute it, +# with or without modifications, as long as this notice is preserved. + +# This program is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY, to the extent permitted by law; without +# even the implied warranty of MERCHANTABILITY or FITNESS FOR A +# PARTICULAR PURPOSE. + +@SET_MAKE@ + + +VPATH = @srcdir@ +pkgdatadir = $(datadir)/@PACKAGE@ +pkgincludedir = $(includedir)/@PACKAGE@ +pkglibdir = $(libdir)/@PACKAGE@ +pkglibexecdir = $(libexecdir)/@PACKAGE@ +am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd +install_sh_DATA = $(install_sh) -c -m 644 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distclean-tags distdir dvi dvi-am html html-am info info-am \ + install install-am install-data install-data-am install-dvi \ + install-dvi-am install-exec install-exec-am install-html \ + install-html-am install-info install-info-am install-man \ + install-pdf install-pdf-am install-pluginLTLIBRARIES \ + install-ps install-ps-am install-strip installcheck \ + installcheck-am installdirs maintainer-clean \ + maintainer-clean-generic mostlyclean mostlyclean-compile \ + mostlyclean-generic mostlyclean-libtool pdf pdf-am ps ps-am \ + tags uninstall uninstall-am uninstall-pluginLTLIBRARIES + + +# Tell versions [3.59,3.63) of GNU make to not export all variables. +# Otherwise a system limit (for SysV at least) may be exceeded. +.NOEXPORT: diff --git a/sys/oss/common.h b/sys/oss/common.h new file mode 100644 index 0000000..2445c3d --- /dev/null +++ b/sys/oss/common.h @@ -0,0 +1,43 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000,2005 Wim Taymans <wim@fluendo.com> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#define SET_PARAM(_oss, _name, _val, _detail) \ +G_STMT_START { \ + int _tmp = _val; \ + if (ioctl(_oss->fd, _name, &_tmp) == -1) { \ + GST_ELEMENT_ERROR (_oss, RESOURCE, SETTINGS,\ + (NULL), \ + ("Unable to set param " _detail ": %s", \ + g_strerror (errno))); \ + return FALSE; \ + } \ + GST_DEBUG_OBJECT(_oss, _detail " %d", _tmp); \ +} G_STMT_END + +#define GET_PARAM(_oss, _name, _val, _detail) \ +G_STMT_START { \ + if (ioctl(oss->fd, _name, _val) == -1) { \ + GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, \ + (NULL), \ + ("Unable to get param " _detail ": %s", \ + g_strerror (errno))); \ + return FALSE; \ + } \ +} G_STMT_END diff --git a/sys/oss/gstossaudio.c b/sys/oss/gstossaudio.c new file mode 100644 index 0000000..2a181e8 --- /dev/null +++ b/sys/oss/gstossaudio.c @@ -0,0 +1,61 @@ +/* GStreamer + * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "gst/gst-i18n-plugin.h" + +#include "gstossmixerelement.h" +#include "gstosssink.h" +#include "gstosssrc.h" + +GST_DEBUG_CATEGORY (oss_debug); +#define GST_CAT_DEFAULT oss_debug + +static gboolean +plugin_init (GstPlugin * plugin) +{ + if (!gst_element_register (plugin, "ossmixer", GST_RANK_NONE, + GST_TYPE_OSS_MIXER_ELEMENT) || + !gst_element_register (plugin, "osssrc", GST_RANK_SECONDARY, + GST_TYPE_OSS_SRC) || + !gst_element_register (plugin, "osssink", GST_RANK_SECONDARY, + GST_TYPE_OSSSINK)) { + return FALSE; + } + + GST_DEBUG_CATEGORY_INIT (oss_debug, "oss", 0, "OSS elements"); + +#ifdef ENABLE_NLS + GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE, + LOCALEDIR); + bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR); + bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8"); +#endif /* ENABLE_NLS */ + + return TRUE; +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "ossaudio", + "OSS (Open Sound System) support for GStreamer", + plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) diff --git a/sys/oss/gstossdmabuffer.h b/sys/oss/gstossdmabuffer.h new file mode 100644 index 0000000..c38ef6f --- /dev/null +++ b/sys/oss/gstossdmabuffer.h @@ -0,0 +1,77 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2005 Wim Taymans <wim@fluendo.com> + * + * gstossdmabuffer.h: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_OSSDMABUFFER_H__ +#define __GST_OSSDMABUFFER_H__ + +#include <gst/gst.h> + +#include "gstosshelper.h" +#include <gst/audio/gstringbuffer.h> + +G_BEGIN_DECLS + +#define GST_TYPE_OSSDMABUFFER (gst_ossdmabuffer_get_type()) +#define GST_OSSDMABUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSDMABUFFER,GstOssDMABuffer)) +#define GST_OSSDMABUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSDMABUFFER,GstOssDMABufferClass)) +#define GST_IS_OSSDMABUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSDMABUFFER)) +#define GST_IS_OSSDMABUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSDMABUFFER)) + +#define GST_OSSELEMENT_GET(obj) GST_OSSELEMENT (obj->element) + +typedef enum { + GST_OSSDMABUFFER_OPEN = (1 << 0), +} GstOssDMABufferFlags; + +typedef struct _GstOssDMABuffer GstOssDMABuffer; +typedef struct _GstOssDMABufferClass GstOssDMABufferClass; + +#define GST_OSSDMABUFFER_THREAD(buf) (GST_OSSDMABUFFER(buf)->thread) +#define GST_OSSDMABUFFER_LOCK GST_OBJECT_LOCK +#define GST_OSSDMABUFFER_UNLOCK GST_OBJECT_UNLOCK +#define GST_OSSDMABUFFER_COND(buf) (GST_OSSDMABUFFER(buf)->cond) +#define GST_OSSDMABUFFER_SIGNAL(buf) (g_cond_signal (GST_OSSDMABUFFER_COND (buf))) +#define GST_OSSDMABUFFER_WAIT(buf) (g_cond_wait (GST_OSSDMABUFFER_COND (buf), GST_OBJECT_GET_LOCK (buf))) + +struct _GstOssDMABuffer { + GstRingBuffer buffer; + + GstOssElement *element; + + int fd; + int caps; + int frag; + + GThread *thread; + GCond *cond; + gboolean running; +}; + +struct _GstOssDMABufferClass { + GstRingBufferClass parent_class; +}; + +GType gst_ossdmabuffer_get_type(void); + +G_END_DECLS + +#endif /* __GST_OSSDMABUFFER_H__ */ diff --git a/sys/oss/gstosshelper.c b/sys/oss/gstosshelper.c new file mode 100644 index 0000000..6d7e6bd --- /dev/null +++ b/sys/oss/gstosshelper.c @@ -0,0 +1,407 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000 Wim Taymans <wim.taymans@chello.be> + * + * gstosshelper.c: OSS helper routines + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "gst/gst-i18n-plugin.h" +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <fcntl.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> + +#ifdef HAVE_OSS_INCLUDE_IN_SYS +# include <sys/soundcard.h> +#else +# ifdef HAVE_OSS_INCLUDE_IN_ROOT +# include <soundcard.h> +# else +# ifdef HAVE_OSS_INCLUDE_IN_MACHINE +# include <machine/soundcard.h> +# else +# error "What to include?" +# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */ +# endif /* HAVE_OSS_INCLUDE_IN_ROOT */ +#endif /* HAVE_OSS_INCLUDE_IN_SYS */ + +#include <gst/interfaces/propertyprobe.h> + +#include "gstosshelper.h" +#include "gstossmixer.h" + +GST_DEBUG_CATEGORY_EXTERN (oss_debug); +#define GST_CAT_DEFAULT oss_debug + +typedef struct _GstOssProbe GstOssProbe; +struct _GstOssProbe +{ + int fd; + int format; + int n_channels; + GArray *rates; + int min; + int max; +}; + +typedef struct _GstOssRange GstOssRange; +struct _GstOssRange +{ + int min; + int max; +}; + +static GstStructure *gst_oss_helper_get_format_structure (unsigned int + format_bit); +static gboolean gst_oss_helper_rate_probe_check (GstOssProbe * probe); +static int gst_oss_helper_rate_check_rate (GstOssProbe * probe, int irate); +static void gst_oss_helper_rate_add_range (GQueue * queue, int min, int max); +static void gst_oss_helper_rate_add_rate (GArray * array, int rate); +static int gst_oss_helper_rate_int_compare (gconstpointer a, gconstpointer b); + +GstCaps * +gst_oss_helper_probe_caps (gint fd) +{ + GstOssProbe *probe; + int i; + gboolean ret; + GstStructure *structure; + unsigned int format_bit; + unsigned int format_mask; + GstCaps *caps; + + /* FIXME test make sure we're not currently playing */ + /* FIXME test both mono and stereo */ + + format_mask = AFMT_U8 | AFMT_S8; + + if (G_BYTE_ORDER == G_LITTLE_ENDIAN) + format_mask |= AFMT_S16_LE | AFMT_U16_LE; + else + format_mask |= AFMT_S16_BE | AFMT_U16_BE; + + caps = gst_caps_new_empty (); + + /* assume that the most significant bit of format_mask is 0 */ + for (format_bit = 1 << 31; format_bit > 0; format_bit >>= 1) { + if (format_bit & format_mask) { + GValue rate_value = { 0 }; + + probe = g_new0 (GstOssProbe, 1); + probe->fd = fd; + probe->format = format_bit; + /* FIXME: this is not working for all cards, see bug #518474 */ + probe->n_channels = 2; + + ret = gst_oss_helper_rate_probe_check (probe); + if (probe->min == -1 || probe->max == -1) { + g_array_free (probe->rates, TRUE); + g_free (probe); + continue; + } + + if (ret) { + GValue value = { 0 }; + + g_array_sort (probe->rates, gst_oss_helper_rate_int_compare); + + g_value_init (&rate_value, GST_TYPE_LIST); + g_value_init (&value, G_TYPE_INT); + + for (i = 0; i < probe->rates->len; i++) { + g_value_set_int (&value, g_array_index (probe->rates, int, i)); + + gst_value_list_append_value (&rate_value, &value); + } + + g_value_unset (&value); + } else { + /* one big range */ + g_value_init (&rate_value, GST_TYPE_INT_RANGE); + gst_value_set_int_range (&rate_value, probe->min, probe->max); + } + + g_array_free (probe->rates, TRUE); + g_free (probe); + + structure = gst_oss_helper_get_format_structure (format_bit); + gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL); + gst_structure_set_value (structure, "rate", &rate_value); + g_value_unset (&rate_value); + + gst_caps_append_structure (caps, structure); + } + } + + if (gst_caps_is_empty (caps)) { + /* fixme: make user-visible */ + GST_WARNING ("Your OSS device could not be probed correctly"); + } + + GST_DEBUG ("probed caps: %" GST_PTR_FORMAT, caps); + + return caps; +} + +static GstStructure * +gst_oss_helper_get_format_structure (unsigned int format_bit) +{ + GstStructure *structure; + int endianness; + gboolean sign; + int width; + + switch (format_bit) { + case AFMT_U8: + endianness = 0; + sign = FALSE; + width = 8; + break; + case AFMT_S16_LE: + endianness = G_LITTLE_ENDIAN; + sign = TRUE; + width = 16; + break; + case AFMT_S16_BE: + endianness = G_BIG_ENDIAN; + sign = TRUE; + width = 16; + break; + case AFMT_S8: + endianness = 0; + sign = TRUE; + width = 8; + break; + case AFMT_U16_LE: + endianness = G_LITTLE_ENDIAN; + sign = FALSE; + width = 16; + break; + case AFMT_U16_BE: + endianness = G_BIG_ENDIAN; + sign = FALSE; + width = 16; + break; + default: + g_assert_not_reached (); + return NULL; + } + + structure = gst_structure_new ("audio/x-raw-int", + "width", G_TYPE_INT, width, + "depth", G_TYPE_INT, width, "signed", G_TYPE_BOOLEAN, sign, NULL); + + if (endianness) { + gst_structure_set (structure, "endianness", G_TYPE_INT, endianness, NULL); + } + + return structure; +} + +static gboolean +gst_oss_helper_rate_probe_check (GstOssProbe * probe) +{ + GstOssRange *range; + GQueue *ranges; + int exact_rates = 0; + gboolean checking_exact_rates = TRUE; + int n_checks = 0; + gboolean result = TRUE; + + ranges = g_queue_new (); + + probe->rates = g_array_new (FALSE, FALSE, sizeof (int)); + + probe->min = gst_oss_helper_rate_check_rate (probe, 1000); + n_checks++; + probe->max = gst_oss_helper_rate_check_rate (probe, 100000); + /* a little bug workaround */ + { + int max; + + max = gst_oss_helper_rate_check_rate (probe, 48000); + if (max > probe->max) { + GST_ERROR + ("Driver bug recognized (driver does not round rates correctly). Please file a bug report."); + probe->max = max; + } + } + n_checks++; + if (probe->min == -1 || probe->max == -1) { + /* This is a workaround for drivers that return -EINVAL (or another + * error) for rates outside of [8000,48000]. If this fails, the + * driver is seriously buggy, and probably doesn't work with other + * media libraries/apps. */ + probe->min = gst_oss_helper_rate_check_rate (probe, 8000); + probe->max = gst_oss_helper_rate_check_rate (probe, 48000); + } + if (probe->min == -1 || probe->max == -1) { + GST_DEBUG ("unexpected check_rate error"); + return FALSE; + } + gst_oss_helper_rate_add_range (ranges, probe->min + 1, probe->max - 1); + + while ((range = g_queue_pop_head (ranges))) { + int min1; + int max1; + int mid; + int mid_ret; + + GST_DEBUG ("checking [%d,%d]", range->min, range->max); + + mid = (range->min + range->max) / 2; + mid_ret = gst_oss_helper_rate_check_rate (probe, mid); + if (mid_ret == -1) { + /* FIXME ioctl returned an error. do something */ + GST_DEBUG ("unexpected check_rate error"); + } + n_checks++; + + if (mid == mid_ret && checking_exact_rates) { + int max_exact_matches = 20; + + exact_rates++; + if (exact_rates > max_exact_matches) { + GST_DEBUG ("got %d exact rates, assuming all are exact", + max_exact_matches); + result = FALSE; + g_free (range); + break; + } + } else { + checking_exact_rates = FALSE; + } + + /* Assume that the rate is arithmetically rounded to the nearest + * supported rate. */ + if (mid == mid_ret) { + min1 = mid - 1; + max1 = mid + 1; + } else { + if (mid < mid_ret) { + min1 = mid - (mid_ret - mid); + max1 = mid_ret + 1; + } else { + min1 = mid_ret - 1; + max1 = mid + (mid - mid_ret); + } + } + + gst_oss_helper_rate_add_range (ranges, range->min, min1); + gst_oss_helper_rate_add_range (ranges, max1, range->max); + + g_free (range); + } + + while ((range = g_queue_pop_head (ranges))) { + g_free (range); + } + g_queue_free (ranges); + + return result; +} + +static void +gst_oss_helper_rate_add_range (GQueue * queue, int min, int max) +{ + if (min <= max) { + GstOssRange *range = g_new0 (GstOssRange, 1); + + range->min = min; + range->max = max; + + g_queue_push_tail (queue, range); + /* push_head also works, but has different probing behavior */ + /*g_queue_push_head (queue, range); */ + } +} + +static int +gst_oss_helper_rate_check_rate (GstOssProbe * probe, int irate) +{ + int rate; + int format; + int n_channels; + int ret; + + rate = irate; + format = probe->format; + n_channels = probe->n_channels; + + GST_LOG ("checking format %d, channels %d, rate %d", + format, n_channels, rate); + ret = ioctl (probe->fd, SNDCTL_DSP_SETFMT, &format); + if (ret < 0 || format != probe->format) { + GST_DEBUG ("unsupported format: %d (%d)", probe->format, format); + return -1; + } + ret = ioctl (probe->fd, SNDCTL_DSP_CHANNELS, &n_channels); + if (ret < 0 || n_channels != probe->n_channels) { + GST_DEBUG ("unsupported channels: %d (%d)", probe->n_channels, n_channels); + return -1; + } + ret = ioctl (probe->fd, SNDCTL_DSP_SPEED, &rate); + if (ret < 0) { + GST_DEBUG ("unsupported rate: %d (%d)", irate, rate); + return -1; + } + + GST_DEBUG ("rate %d -> %d", irate, rate); + + if (rate == irate - 1 || rate == irate + 1) { + rate = irate; + } + gst_oss_helper_rate_add_rate (probe->rates, rate); + return rate; +} + +static void +gst_oss_helper_rate_add_rate (GArray * array, int rate) +{ + int i; + int val; + + for (i = 0; i < array->len; i++) { + val = g_array_index (array, int, i); + + if (val == rate) + return; + } + GST_DEBUG ("supported rate: %d", rate); + g_array_append_val (array, rate); +} + +static int +gst_oss_helper_rate_int_compare (gconstpointer a, gconstpointer b) +{ + const int *va = (const int *) a; + const int *vb = (const int *) b; + + if (*va < *vb) + return -1; + if (*va > *vb) + return 1; + return 0; +} diff --git a/sys/oss/gstosshelper.h b/sys/oss/gstosshelper.h new file mode 100644 index 0000000..1593fe3 --- /dev/null +++ b/sys/oss/gstosshelper.h @@ -0,0 +1,43 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000 Wim Taymans <wim.taymans@chello.be> + * + * gstosshelper.h: OSS helper routines. + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#ifndef __GST_OSS_HELPER_H__ +#define __GST_OSS_HELPER_H__ + + +#include <gst/gst.h> +#include <sys/types.h> + +#include "gstosshelper.h" + + +G_BEGIN_DECLS + + +GstCaps* gst_oss_helper_probe_caps (gint fd); + + +G_END_DECLS + + +#endif /* __GST_OSS_HELPER_H__ */ diff --git a/sys/oss/gstossmixer.c b/sys/oss/gstossmixer.c new file mode 100644 index 0000000..e563031 --- /dev/null +++ b/sys/oss/gstossmixer.c @@ -0,0 +1,378 @@ +/* GStreamer OSS Mixer implementation + * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net> + * + * gstossmixer.c: mixer interface implementation for OSS + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stdio.h> +#include <stdlib.h> +#include <fcntl.h> +#include <unistd.h> +#include <string.h> +#include <errno.h> +#include <sys/ioctl.h> + +#ifdef HAVE_OSS_INCLUDE_IN_SYS +# include <sys/soundcard.h> +#else +# ifdef HAVE_OSS_INCLUDE_IN_ROOT +# include <soundcard.h> +# else +# ifdef HAVE_OSS_INCLUDE_IN_MACHINE +# include <machine/soundcard.h> +# else +# error "What to include?" +# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */ +# endif /* HAVE_OSS_INCLUDE_IN_ROOT */ +#endif /* HAVE_OSS_INCLUDE_IN_SYS */ + +#include <gst/gst-i18n-plugin.h> + +#include "gstossmixer.h" +#include "gstossmixertrack.h" + +GST_DEBUG_CATEGORY_EXTERN (oss_debug); +#define GST_CAT_DEFAULT oss_debug + +#define MASK_BIT_IS_SET(mask, bit) \ + (mask & (1 << bit)) + +static gboolean +gst_ossmixer_open (GstOssMixer * mixer) +{ +#ifdef SOUND_MIXER_INFO + struct mixer_info minfo; +#endif + + g_return_val_if_fail (mixer->mixer_fd == -1, FALSE); + + mixer->mixer_fd = open (mixer->device, O_RDWR); + if (mixer->mixer_fd == -1) + goto open_failed; + + /* get masks */ + if (ioctl (mixer->mixer_fd, SOUND_MIXER_READ_RECMASK, &mixer->recmask) < 0 + || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_RECSRC, &mixer->recdevs) < 0 + || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_STEREODEVS, + &mixer->stereomask) < 0 + || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_DEVMASK, &mixer->devmask) < 0 + || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_CAPS, &mixer->mixcaps) < 0) + goto masks_failed; + + /* get name, not fatal */ + g_free (mixer->cardname); +#ifdef SOUND_MIXER_INFO + if (ioctl (mixer->mixer_fd, SOUND_MIXER_INFO, &minfo) == 0) { + mixer->cardname = g_strdup (minfo.name); + GST_INFO ("Card name = %s", GST_STR_NULL (mixer->cardname)); + } else +#endif + { + mixer->cardname = g_strdup ("Unknown"); + GST_INFO ("Unknown card name"); + } + GST_INFO ("Opened mixer for device %s", mixer->device); + + return TRUE; + + /* ERRORS */ +open_failed: + { + /* this is valid. OSS devices don't need to expose a mixer */ + GST_DEBUG ("Failed to open mixer device %s, mixing disabled: %s", + mixer->device, strerror (errno)); + return FALSE; + } +masks_failed: + { + GST_DEBUG ("Failed to get device masks"); + close (mixer->mixer_fd); + mixer->mixer_fd = -1; + return FALSE; + } +} + +static void +gst_ossmixer_ensure_track_list (GstOssMixer * mixer) +{ + gint i, master = -1; + + g_return_if_fail (mixer->mixer_fd != -1); + + if (mixer->tracklist) + return; + + /* find master volume */ + if (mixer->devmask & SOUND_MASK_VOLUME) + master = SOUND_MIXER_VOLUME; + else if (mixer->devmask & SOUND_MASK_PCM) + master = SOUND_MIXER_PCM; + else if (mixer->devmask & SOUND_MASK_SPEAKER) + master = SOUND_MIXER_SPEAKER; /* doubtful... */ + /* else: no master, so we won't set any */ + + /* build track list */ + for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) { + if (mixer->devmask & (1 << i)) { + GstMixerTrack *track; + gboolean input = FALSE, stereo = FALSE, record = FALSE; + + /* track exists, make up capabilities */ + if (MASK_BIT_IS_SET (mixer->stereomask, i)) + stereo = TRUE; + if (MASK_BIT_IS_SET (mixer->recmask, i)) + input = TRUE; + if (MASK_BIT_IS_SET (mixer->recdevs, i)) + record = TRUE; + + /* do we want this in our list? */ + if (!((mixer->dir & GST_OSS_MIXER_CAPTURE && input == TRUE) || + (mixer->dir & GST_OSS_MIXER_PLAYBACK && i != SOUND_MIXER_PCM))) + /* the PLAYBACK case seems hacky, but that's how 0.8 had it */ + continue; + + /* add track to list */ + track = gst_ossmixer_track_new (mixer->mixer_fd, i, stereo ? 2 : 1, + (record ? GST_MIXER_TRACK_RECORD : 0) | + (input ? GST_MIXER_TRACK_INPUT : + GST_MIXER_TRACK_OUTPUT) | + ((master != i) ? 0 : GST_MIXER_TRACK_MASTER)); + mixer->tracklist = g_list_append (mixer->tracklist, track); + } + } +} + +GstOssMixer * +gst_ossmixer_new (const char *device, GstOssMixerDirection dir) +{ + GstOssMixer *ret = NULL; + + g_return_val_if_fail (device != NULL, NULL); + + ret = g_new0 (GstOssMixer, 1); + + ret->device = g_strdup (device); + ret->dir = dir; + ret->mixer_fd = -1; + + if (!gst_ossmixer_open (ret)) + goto error; + + return ret; + + /* ERRORS */ +error: + { + gst_ossmixer_free (ret); + return NULL; + } +} + +void +gst_ossmixer_free (GstOssMixer * mixer) +{ + g_return_if_fail (mixer != NULL); + + if (mixer->device) { + g_free (mixer->device); + mixer->device = NULL; + } + + if (mixer->cardname) { + g_free (mixer->cardname); + mixer->cardname = NULL; + } + + if (mixer->tracklist) { + g_list_foreach (mixer->tracklist, (GFunc) g_object_unref, NULL); + g_list_free (mixer->tracklist); + mixer->tracklist = NULL; + } + + if (mixer->mixer_fd != -1) { + close (mixer->mixer_fd); + mixer->mixer_fd = -1; + } + + g_free (mixer); +} + +/* unused with G_DISABLE_* */ +static G_GNUC_UNUSED gboolean +gst_ossmixer_contains_track (GstOssMixer * mixer, GstOssMixerTrack * osstrack) +{ + const GList *item; + + for (item = mixer->tracklist; item != NULL; item = item->next) + if (item->data == osstrack) + return TRUE; + + return FALSE; +} + +const GList * +gst_ossmixer_list_tracks (GstOssMixer * mixer) +{ + gst_ossmixer_ensure_track_list (mixer); + + return (const GList *) mixer->tracklist; +} + +void +gst_ossmixer_get_volume (GstOssMixer * mixer, + GstMixerTrack * track, gint * volumes) +{ + gint volume; + GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track); + + g_return_if_fail (mixer->mixer_fd != -1); + g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack)); + + if (track->flags & GST_MIXER_TRACK_MUTE) { + volumes[0] = osstrack->lvol; + if (track->num_channels == 2) { + volumes[1] = osstrack->rvol; + } + } else { + /* get */ + if (ioctl (mixer->mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) { + g_warning ("Error getting recording device (%d) volume: %s", + osstrack->track_num, strerror (errno)); + volume = 0; + } + + osstrack->lvol = volumes[0] = (volume & 0xff); + if (track->num_channels == 2) { + osstrack->rvol = volumes[1] = ((volume >> 8) & 0xff); + } + } +} + +void +gst_ossmixer_set_volume (GstOssMixer * mixer, + GstMixerTrack * track, gint * volumes) +{ + gint volume; + GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track); + + g_return_if_fail (mixer->mixer_fd != -1); + g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack)); + + /* prepare the value for ioctl() */ + if (!(track->flags & GST_MIXER_TRACK_MUTE)) { + volume = (volumes[0] & 0xff); + if (track->num_channels == 2) { + volume |= ((volumes[1] & 0xff) << 8); + } + + /* set */ + if (ioctl (mixer->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) { + g_warning ("Error setting recording device (%d) volume (0x%x): %s", + osstrack->track_num, volume, strerror (errno)); + return; + } + } + + osstrack->lvol = volumes[0]; + if (track->num_channels == 2) { + osstrack->rvol = volumes[1]; + } +} + +void +gst_ossmixer_set_mute (GstOssMixer * mixer, GstMixerTrack * track, + gboolean mute) +{ + int volume; + GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track); + + g_return_if_fail (mixer->mixer_fd != -1); + g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack)); + + if (mute) { + volume = 0; + } else { + volume = (osstrack->lvol & 0xff); + if (MASK_BIT_IS_SET (mixer->stereomask, osstrack->track_num)) { + volume |= ((osstrack->rvol & 0xff) << 8); + } + } + + if (ioctl (mixer->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) { + g_warning ("Error setting mixer recording device volume (0x%x): %s", + volume, strerror (errno)); + return; + } + + if (mute) { + track->flags |= GST_MIXER_TRACK_MUTE; + } else { + track->flags &= ~GST_MIXER_TRACK_MUTE; + } +} + +void +gst_ossmixer_set_record (GstOssMixer * mixer, + GstMixerTrack * track, gboolean record) +{ + GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track); + + g_return_if_fail (mixer->mixer_fd != -1); + g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack)); + + /* if there's nothing to do... */ + if ((record && GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) || + (!record && !GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD))) + return; + + /* if we're exclusive, then we need to unset the current one(s) */ + if (mixer->mixcaps & SOUND_CAP_EXCL_INPUT) { + GList *track; + + for (track = mixer->tracklist; track != NULL; track = track->next) { + GstMixerTrack *turn = (GstMixerTrack *) track->data; + + turn->flags &= ~GST_MIXER_TRACK_RECORD; + } + mixer->recdevs = 0; + } + + /* set new record bit, if needed */ + if (record) { + mixer->recdevs |= (1 << osstrack->track_num); + } else { + mixer->recdevs &= ~(1 << osstrack->track_num); + } + + /* set it to the device */ + if (ioctl (mixer->mixer_fd, SOUND_MIXER_WRITE_RECSRC, &mixer->recdevs) < 0) { + g_warning ("Error setting mixer recording devices (0x%x): %s", + mixer->recdevs, strerror (errno)); + return; + } + + if (record) { + track->flags |= GST_MIXER_TRACK_RECORD; + } else { + track->flags &= ~GST_MIXER_TRACK_RECORD; + } +} diff --git a/sys/oss/gstossmixer.h b/sys/oss/gstossmixer.h new file mode 100644 index 0000000..d2e06fe --- /dev/null +++ b/sys/oss/gstossmixer.h @@ -0,0 +1,171 @@ +/* GStreamer OSS Mixer implementation + * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net> + * + * gstossmixer.h: mixer interface implementation for OSS + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#ifndef __GST_OSS_MIXER_H__ +#define __GST_OSS_MIXER_H__ + + +#include <gst/gst.h> +#include <gst/interfaces/mixer.h> + +#include "gstosshelper.h" + + +G_BEGIN_DECLS + + +#define GST_OSS_MIXER(obj) ((GstOssMixer*)(obj)) + + +typedef enum { + GST_OSS_MIXER_CAPTURE = 1<<0, + GST_OSS_MIXER_PLAYBACK = 1<<1, + GST_OSS_MIXER_ALL = GST_OSS_MIXER_CAPTURE | GST_OSS_MIXER_PLAYBACK +} GstOssMixerDirection; + + +typedef struct _GstOssMixer GstOssMixer; + + +struct _GstOssMixer { + GList * tracklist; /* list of available tracks */ + + gint mixer_fd; + + gchar * device; + gchar * cardname; + + gint recmask; + gint recdevs; + gint stereomask; + gint devmask; + gint mixcaps; + + GstOssMixerDirection dir; +}; + + +GstOssMixer* gst_ossmixer_new (const gchar *device, + GstOssMixerDirection dir); +void gst_ossmixer_free (GstOssMixer *mixer); + +const GList* gst_ossmixer_list_tracks (GstOssMixer * mixer); +void gst_ossmixer_set_volume (GstOssMixer * mixer, + GstMixerTrack * track, + gint * volumes); +void gst_ossmixer_get_volume (GstOssMixer * mixer, + GstMixerTrack * track, + gint * volumes); +void gst_ossmixer_set_record (GstOssMixer * mixer, + GstMixerTrack * track, + gboolean record); +void gst_ossmixer_set_mute (GstOssMixer * mixer, + GstMixerTrack * track, + gboolean mute); + + +#define GST_IMPLEMENT_OSS_MIXER_METHODS(Type, interface_as_function) \ +static gboolean \ +interface_as_function ## _supported (Type *this, GType iface_type) \ +{ \ + g_assert (iface_type == GST_TYPE_MIXER); \ + \ + return (this->mixer != NULL); \ +} \ + \ +static const GList* \ +interface_as_function ## _list_tracks (GstMixer * mixer) \ +{ \ + Type *this = (Type*) mixer; \ + \ + g_return_val_if_fail (this != NULL, NULL); \ + g_return_val_if_fail (this->mixer != NULL, NULL); \ + \ + return gst_ossmixer_list_tracks (this->mixer); \ +} \ + \ +static void \ +interface_as_function ## _set_volume (GstMixer * mixer, GstMixerTrack * track, \ + gint * volumes) \ +{ \ + Type *this = (Type*) mixer; \ + \ + g_return_if_fail (this != NULL); \ + g_return_if_fail (this->mixer != NULL); \ + \ + gst_ossmixer_set_volume (this->mixer, track, volumes); \ +} \ + \ +static void \ +interface_as_function ## _get_volume (GstMixer * mixer, GstMixerTrack * track, \ + gint * volumes) \ +{ \ + Type *this = (Type*) mixer; \ + \ + g_return_if_fail (this != NULL); \ + g_return_if_fail (this->mixer != NULL); \ + \ + gst_ossmixer_get_volume (this->mixer, track, volumes); \ +} \ + \ +static void \ +interface_as_function ## _set_record (GstMixer * mixer, GstMixerTrack * track, \ + gboolean record) \ +{ \ + Type *this = (Type*) mixer; \ + \ + g_return_if_fail (this != NULL); \ + g_return_if_fail (this->mixer != NULL); \ + \ + gst_ossmixer_set_record (this->mixer, track, record); \ +} \ + \ +static void \ +interface_as_function ## _set_mute (GstMixer * mixer, GstMixerTrack * track, \ + gboolean mute) \ +{ \ + Type *this = (Type*) mixer; \ + \ + g_return_if_fail (this != NULL); \ + g_return_if_fail (this->mixer != NULL); \ + \ + gst_ossmixer_set_mute (this->mixer, track, mute); \ +} \ + \ +static void \ +interface_as_function ## _interface_init (GstMixerClass * klass) \ +{ \ + GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE; \ + \ + /* set up the interface hooks */ \ + klass->list_tracks = interface_as_function ## _list_tracks; \ + klass->set_volume = interface_as_function ## _set_volume; \ + klass->get_volume = interface_as_function ## _get_volume; \ + klass->set_mute = interface_as_function ## _set_mute; \ + klass->set_record = interface_as_function ## _set_record; \ +} + + +G_END_DECLS + + +#endif /* __GST_OSS_MIXER_H__ */ diff --git a/sys/oss/gstossmixerelement.c b/sys/oss/gstossmixerelement.c new file mode 100644 index 0000000..7ceb2b8 --- /dev/null +++ b/sys/oss/gstossmixerelement.c @@ -0,0 +1,218 @@ +/* OSS mixer interface element. + * Copyright (C) 2005 Andrew Vander Wingo <wingo@pobox.com> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-ossmixer + * + * This element lets you adjust sound input and output levels with the + * Open Sound System (OSS). It supports the #GstMixer interface, which can be + * used to obtain a list of available mixer tracks. Set the mixer element to + * READY state before using the #GstMixer interface on it. + * + * <refsect2> + * <title>Example pipelines</title> + * <para> + * ossmixer can't be used in a sensible way in gst-launch. + * </para> + * </refsect2> + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include "gstossmixerelement.h" + +GST_DEBUG_CATEGORY_EXTERN (oss_debug); +#define GST_CAT_DEFAULT oss_debug + +#define DEFAULT_DEVICE "/dev/mixer" +#define DEFAULT_DEVICE_NAME NULL + +enum +{ + PROP_0, + PROP_DEVICE, + PROP_DEVICE_NAME +}; + +GST_BOILERPLATE_WITH_INTERFACE (GstOssMixerElement, gst_oss_mixer_element, + GstElement, GST_TYPE_ELEMENT, GstMixer, GST_TYPE_MIXER, + gst_oss_mixer_element); + +GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssMixerElement, gst_oss_mixer_element); + +static GstStateChangeReturn gst_oss_mixer_element_change_state (GstElement * + element, GstStateChange transition); + +static void gst_oss_mixer_element_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_oss_mixer_element_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); +static void gst_oss_mixer_element_finalize (GObject * object); + +static void +gst_oss_mixer_element_base_init (gpointer klass) +{ + gst_element_class_set_details_simple (GST_ELEMENT_CLASS (klass), "OSS Mixer", + "Generic/Audio", + "Control sound input and output levels with OSS", + "Andrew Vander Wingo <wingo@pobox.com>"); +} + +static void +gst_oss_mixer_element_class_init (GstOssMixerElementClass * klass) +{ + GstElementClass *element_class; + GObjectClass *gobject_class; + + element_class = (GstElementClass *) klass; + gobject_class = (GObjectClass *) klass; + + gobject_class->finalize = gst_oss_mixer_element_finalize; + gobject_class->set_property = gst_oss_mixer_element_set_property; + gobject_class->get_property = gst_oss_mixer_element_get_property; + + /** + * GstOssMixerElement:device + * + * OSS mixer device (usually /dev/mixer) + * + * Since: 0.10.5 + **/ + g_object_class_install_property (gobject_class, PROP_DEVICE, + g_param_spec_string ("device", "Device", + "OSS mixer device (usually /dev/mixer)", DEFAULT_DEVICE, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_DEVICE_NAME, + g_param_spec_string ("device-name", "Device name", + "Human-readable name of the sound device", DEFAULT_DEVICE_NAME, + G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); + + element_class->change_state = + GST_DEBUG_FUNCPTR (gst_oss_mixer_element_change_state); +} + +static void +gst_oss_mixer_element_finalize (GObject * obj) +{ + GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (obj); + + g_free (this->device); + + G_OBJECT_CLASS (parent_class)->finalize (obj); +} + +static void +gst_oss_mixer_element_init (GstOssMixerElement * this, + GstOssMixerElementClass * g_class) +{ + this->mixer = NULL; + this->device = g_strdup (DEFAULT_DEVICE); +} + +static void +gst_oss_mixer_element_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (object); + + switch (prop_id) { + case PROP_DEVICE: + g_free (this->device); + this->device = g_value_dup_string (value); + /* make sure we never set NULL */ + if (this->device == NULL) { + this->device = g_strdup (DEFAULT_DEVICE); + } + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_oss_mixer_element_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (object); + + switch (prop_id) { + case PROP_DEVICE: + g_value_set_string (value, this->device); + break; + case PROP_DEVICE_NAME: + if (this->mixer) { + g_value_set_string (value, this->mixer->cardname); + } else { + g_value_set_string (value, NULL); + } + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstStateChangeReturn +gst_oss_mixer_element_change_state (GstElement * element, + GstStateChange transition) +{ + GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; + GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (element); + + switch (transition) { + case GST_STATE_CHANGE_NULL_TO_READY: + if (!this->mixer) { + this->mixer = gst_ossmixer_new (this->device, GST_OSS_MIXER_ALL); + if (!this->mixer) + goto open_failed; + } + break; + break; + default: + break; + } + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + if (ret == GST_STATE_CHANGE_FAILURE) + return ret; + + switch (transition) { + case GST_STATE_CHANGE_READY_TO_NULL: + if (this->mixer) { + gst_ossmixer_free (this->mixer); + this->mixer = NULL; + } + break; + default: + break; + } + return ret; + + /* ERRORS */ +open_failed: + { + GST_ELEMENT_ERROR (element, RESOURCE, OPEN_READ_WRITE, (NULL), + ("Failed to open oss mixer device '%s'", this->device)); + return GST_STATE_CHANGE_FAILURE; + } +} diff --git a/sys/oss/gstossmixerelement.h b/sys/oss/gstossmixerelement.h new file mode 100644 index 0000000..2d47e0e --- /dev/null +++ b/sys/oss/gstossmixerelement.h @@ -0,0 +1,59 @@ +/* OSS mixer interface element. + * Copyright (C) 2005 Andrew Vander Wingo <wingo@pobox.com> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the Free + * Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + + +#ifndef __GST_OSS_MIXER_ELEMENT_H__ +#define __GST_OSS_MIXER_ELEMENT_H__ + + +#include "gstossmixer.h" + + +G_BEGIN_DECLS + + +#define GST_OSS_MIXER_ELEMENT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS_MIXER_ELEMENT,GstOssMixerElement)) +#define GST_OSS_MIXER_ELEMENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS_MIXER_ELEMENT,GstOssMixerElementClass)) +#define GST_IS_OSS_MIXER_ELEMENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS_MIXER_ELEMENT)) +#define GST_IS_OSS_MIXER_ELEMENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS_MIXER_ELEMENT)) +#define GST_TYPE_OSS_MIXER_ELEMENT (gst_oss_mixer_element_get_type()) + + +typedef struct _GstOssMixerElement GstOssMixerElement; +typedef struct _GstOssMixerElementClass GstOssMixerElementClass; + + +struct _GstOssMixerElement { + GstElement parent; + + gchar *device; + GstOssMixer *mixer; +}; + +struct _GstOssMixerElementClass { + GstElementClass parent; +}; + + +GType gst_oss_mixer_element_get_type (void); + + +G_END_DECLS + + +#endif /* __GST_OSS_MIXER_ELEMENT_H__ */ diff --git a/sys/oss/gstossmixertrack.c b/sys/oss/gstossmixertrack.c new file mode 100644 index 0000000..3bc8cde --- /dev/null +++ b/sys/oss/gstossmixertrack.c @@ -0,0 +1,172 @@ +/* GStreamer OSS Mixer implementation + * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net> + * + * gstossmixer.c: mixer interface implementation for OSS + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stdio.h> +#include <stdlib.h> +#include <fcntl.h> +#include <unistd.h> +#include <string.h> +#include <errno.h> +#include <sys/ioctl.h> + +#ifdef HAVE_OSS_INCLUDE_IN_SYS +# include <sys/soundcard.h> +#else +# ifdef HAVE_OSS_INCLUDE_IN_ROOT +# include <soundcard.h> +# else +# ifdef HAVE_OSS_INCLUDE_IN_MACHINE +# include <machine/soundcard.h> +# else +# error "What to include?" +# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */ +# endif /* HAVE_OSS_INCLUDE_IN_ROOT */ +#endif /* HAVE_OSS_INCLUDE_IN_SYS */ + +#include <gst/gst-i18n-plugin.h> + +#include "gstossmixertrack.h" + +GST_DEBUG_CATEGORY_EXTERN (oss_debug); +#define GST_CAT_DEFAULT oss_debug + +#define MASK_BIT_IS_SET(mask, bit) \ + (mask & (1 << bit)) + +G_DEFINE_TYPE (GstOssMixerTrack, gst_ossmixer_track, GST_TYPE_MIXER_TRACK); + +static void +gst_ossmixer_track_class_init (GstOssMixerTrackClass * klass) +{ + /* nop */ +} + +static void +gst_ossmixer_track_init (GstOssMixerTrack * track) +{ + track->lvol = track->rvol = 0; + track->track_num = 0; +} + +static const gchar **labels = NULL; + +/* three functions: firstly, OSS has the nasty habit of inserting + * spaces in the labels, we want to get rid of them. Secondly, + * i18n is impossible with OSS' way of providing us with mixer + * labels, so we make a 'given' list of i18n'ed labels. Thirdly, I + * personally don't like the "1337" names that OSS gives to their + * labels ("Vol", "Mic", "Rec"), I'd rather see full names. */ + +static void +fill_labels (void) +{ + gint i, pos; + const gchar *origs[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_LABELS; + const struct + { + const gchar *given; + const gchar *wanted; + } + cases[] = { + /* Note: this list is simply ripped from soundcard.h. For + * some people, some values might be missing (3D surround, + * etc.) - feel free to add them. That's the reason why + * I'm doing this in such a horribly complicated way. */ + { + "Vol ", _("Volume")}, { + "Bass ", _("Bass")}, { + "Trebl", _("Treble")}, { + "Synth", _("Synth")}, { + "Pcm ", _("PCM")}, { + "Spkr ", _("Speaker")}, { + "Line ", _("Line-in")}, { + "Mic ", _("Microphone")}, { + "CD ", _("CD")}, { + "Mix ", _("Mixer")}, { + "Pcm2 ", _("PCM-2")}, { + "Rec ", _("Record")}, { + "IGain", _("In-gain")}, { + "OGain", _("Out-gain")}, { + "Line1", _("Line-1")}, { + "Line2", _("Line-2")}, { + "Line3", _("Line-3")}, { + "Digital1", _("Digital-1")}, { + "Digital2", _("Digital-2")}, { + "Digital3", _("Digital-3")}, { + "PhoneIn", _("Phone-in")}, { + "PhoneOut", _("Phone-out")}, { + "Video", _("Video")}, { + "Radio", _("Radio")}, { + "Monitor", _("Monitor")}, { + NULL, NULL} + }; + + labels = g_malloc (sizeof (gchar *) * SOUND_MIXER_NRDEVICES); + + for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) { + for (pos = 0; cases[pos].given != NULL; pos++) { + if (!strcmp (cases[pos].given, origs[i])) { + labels[i] = g_strdup (cases[pos].wanted); + break; + } + } + if (cases[pos].given == NULL) + labels[i] = g_strdup (origs[i]); + } +} + +GstMixerTrack * +gst_ossmixer_track_new (gint mixer_fd, + gint track_num, gint max_chans, gint flags) +{ + GstOssMixerTrack *osstrack; + GstMixerTrack *track; + gint volume; + + if (!labels) + fill_labels (); + + osstrack = g_object_new (GST_TYPE_OSSMIXER_TRACK, NULL); + track = GST_MIXER_TRACK (osstrack); + track->label = g_strdup (labels[track_num]); + track->num_channels = max_chans; + track->flags = flags; + track->min_volume = 0; + track->max_volume = 100; + osstrack->track_num = track_num; + + /* volume */ + if (ioctl (mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) { + g_warning ("Error getting device (%d) volume: %s", + osstrack->track_num, strerror (errno)); + volume = 0; + } + osstrack->lvol = (volume & 0xff); + if (track->num_channels == 2) { + osstrack->rvol = ((volume >> 8) & 0xff); + } + + return track; +} diff --git a/sys/oss/gstossmixertrack.h b/sys/oss/gstossmixertrack.h new file mode 100644 index 0000000..2e62ec8 --- /dev/null +++ b/sys/oss/gstossmixertrack.h @@ -0,0 +1,62 @@ +/* GStreamer OSS Mixer implementation + * Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net> + * + * gstossmixertrack.h: OSS mixer tracks + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_OSS_MIXER_TRACK_H__ +#define __GST_OSS_MIXER_TRACK_H__ + +#include <gst/gst.h> +#include <gst/interfaces/mixer.h> + +#include "gstosshelper.h" + +G_BEGIN_DECLS + +#define GST_TYPE_OSSMIXER_TRACK \ + (gst_ossmixer_track_get_type ()) +#define GST_OSSMIXER_TRACK(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_OSSMIXER_TRACK, \ + GstOssMixerTrack)) +#define GST_OSSMIXER_TRACK_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_OSSMIXER_TRACK, \ + GstOssMixerTrackClass)) +#define GST_IS_OSSMIXER_TRACK(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_OSSMIXER_TRACK)) +#define GST_IS_OSSMIXER_TRACK_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_OSSMIXER_TRACK)) + +typedef struct _GstOssMixerTrack { + GstMixerTrack parent; + + gint lvol, rvol; + gint track_num; +} GstOssMixerTrack; + +typedef struct _GstOssMixerTrackClass { + GstMixerTrackClass parent; +} GstOssMixerTrackClass; + +GType gst_ossmixer_track_get_type (void); +GstMixerTrack* gst_ossmixer_track_new (gint mixer_fd, + gint track_num, gint max_chans, gint flags); + +G_END_DECLS + +#endif /* __GST_OSS_MIXER_TRACK_H__ */ diff --git a/sys/oss/gstosssink.c b/sys/oss/gstosssink.c new file mode 100644 index 0000000..9f9b137 --- /dev/null +++ b/sys/oss/gstosssink.c @@ -0,0 +1,565 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000,2005 Wim Taymans <wim@fluendo.com> + * + * gstosssink.c: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-osssink + * + * This element lets you output sound using the Open Sound System (OSS). + * + * Note that you should almost always use generic audio conversion elements + * like audioconvert and audioresample in front of an audiosink to make sure + * your pipeline works under all circumstances (those conversion elements will + * act in passthrough-mode if no conversion is necessary). + * + * <refsect2> + * <title>Example pipelines</title> + * |[ + * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! osssink + * ]| will output a sine wave (continuous beep sound) to your sound card (with + * a very low volume as precaution). + * |[ + * gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! osssink + * ]| will play an Ogg/Vorbis audio file and output it using the Open Sound System. + * </refsect2> + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include <sys/ioctl.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <string.h> + +#ifdef HAVE_OSS_INCLUDE_IN_SYS +# include <sys/soundcard.h> +#else +# ifdef HAVE_OSS_INCLUDE_IN_ROOT +# include <soundcard.h> +# else +# ifdef HAVE_OSS_INCLUDE_IN_MACHINE +# include <machine/soundcard.h> +# else +# error "What to include?" +# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */ +# endif /* HAVE_OSS_INCLUDE_IN_ROOT */ +#endif /* HAVE_OSS_INCLUDE_IN_SYS */ + +#include "common.h" +#include "gstosssink.h" + +#include <gst/gst-i18n-plugin.h> + +GST_DEBUG_CATEGORY_EXTERN (oss_debug); +#define GST_CAT_DEFAULT oss_debug + +static void gst_oss_sink_base_init (gpointer g_class); +static void gst_oss_sink_class_init (GstOssSinkClass * klass); +static void gst_oss_sink_init (GstOssSink * osssink); + +static void gst_oss_sink_dispose (GObject * object); +static void gst_oss_sink_finalise (GObject * object); + +static void gst_oss_sink_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static void gst_oss_sink_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); + +static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink); + +static gboolean gst_oss_sink_open (GstAudioSink * asink); +static gboolean gst_oss_sink_close (GstAudioSink * asink); +static gboolean gst_oss_sink_prepare (GstAudioSink * asink, + GstRingBufferSpec * spec); +static gboolean gst_oss_sink_unprepare (GstAudioSink * asink); +static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data, + guint length); +static guint gst_oss_sink_delay (GstAudioSink * asink); +static void gst_oss_sink_reset (GstAudioSink * asink); + +/* OssSink signals and args */ +enum +{ + LAST_SIGNAL +}; + +#define DEFAULT_DEVICE "/dev/dsp" +enum +{ + PROP_0, + PROP_DEVICE, +}; + +static GstStaticPadTemplate osssink_sink_factory = + GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-int, " + "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, " + "signed = (boolean) { TRUE, FALSE }, " + "width = (int) 16, " + "depth = (int) 16, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; " + "audio/x-raw-int, " + "signed = (boolean) { TRUE, FALSE }, " + "width = (int) 8, " + "depth = (int) 8, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]") + ); + +static GstElementClass *parent_class = NULL; + +/* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */ + +GType +gst_oss_sink_get_type (void) +{ + static GType osssink_type = 0; + + if (!osssink_type) { + static const GTypeInfo osssink_info = { + sizeof (GstOssSinkClass), + gst_oss_sink_base_init, + NULL, + (GClassInitFunc) gst_oss_sink_class_init, + NULL, + NULL, + sizeof (GstOssSink), + 0, + (GInstanceInitFunc) gst_oss_sink_init, + }; + + osssink_type = + g_type_register_static (GST_TYPE_AUDIO_SINK, "GstOssSink", + &osssink_info, 0); + } + + return osssink_type; +} + +static void +gst_oss_sink_dispose (GObject * object) +{ + GstOssSink *osssink = GST_OSSSINK (object); + + if (osssink->probed_caps) { + gst_caps_unref (osssink->probed_caps); + osssink->probed_caps = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_oss_sink_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_set_details_simple (element_class, "Audio Sink (OSS)", + "Sink/Audio", + "Output to a sound card via OSS", + "Erik Walthinsen <omega@cse.ogi.edu>, " + "Wim Taymans <wim.taymans@chello.be>"); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&osssink_sink_factory)); +} + +static void +gst_oss_sink_class_init (GstOssSinkClass * klass) +{ + GObjectClass *gobject_class; + GstBaseSinkClass *gstbasesink_class; + GstAudioSinkClass *gstaudiosink_class; + + gobject_class = (GObjectClass *) klass; + gstbasesink_class = (GstBaseSinkClass *) klass; + gstaudiosink_class = (GstAudioSinkClass *) klass; + + parent_class = g_type_class_peek_parent (klass); + + gobject_class->dispose = gst_oss_sink_dispose; + gobject_class->finalize = gst_oss_sink_finalise; + gobject_class->get_property = gst_oss_sink_get_property; + gobject_class->set_property = gst_oss_sink_set_property; + + g_object_class_install_property (gobject_class, PROP_DEVICE, + g_param_spec_string ("device", "Device", + "OSS device (usually /dev/dspN)", DEFAULT_DEVICE, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps); + + gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open); + gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close); + gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare); + gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare); + gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write); + gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay); + gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset); +} + +static void +gst_oss_sink_init (GstOssSink * osssink) +{ + const gchar *device; + + GST_DEBUG_OBJECT (osssink, "initializing osssink"); + + device = g_getenv ("AUDIODEV"); + if (device == NULL) + device = DEFAULT_DEVICE; + osssink->device = g_strdup (device); + osssink->fd = -1; +} + +static void +gst_oss_sink_finalise (GObject * object) +{ + GstOssSink *osssink = GST_OSSSINK (object); + + g_free (osssink->device); + + G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (object)); +} + +static void +gst_oss_sink_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstOssSink *sink; + + sink = GST_OSSSINK (object); + + switch (prop_id) { + case PROP_DEVICE: + g_free (sink->device); + sink->device = g_value_dup_string (value); + if (sink->probed_caps) { + gst_caps_unref (sink->probed_caps); + sink->probed_caps = NULL; + } + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_oss_sink_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstOssSink *sink; + + sink = GST_OSSSINK (object); + + switch (prop_id) { + case PROP_DEVICE: + g_value_set_string (value, sink->device); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstCaps * +gst_oss_sink_getcaps (GstBaseSink * bsink) +{ + GstOssSink *osssink; + GstCaps *caps; + + osssink = GST_OSSSINK (bsink); + + if (osssink->fd == -1) { + caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD + (bsink))); + } else if (osssink->probed_caps) { + caps = gst_caps_copy (osssink->probed_caps); + } else { + caps = gst_oss_helper_probe_caps (osssink->fd); + if (caps && !gst_caps_is_empty (caps)) { + osssink->probed_caps = gst_caps_copy (caps); + } + } + + return caps; +} + +static gint +ilog2 (gint x) +{ + /* well... hacker's delight explains... */ + x = x | (x >> 1); + x = x | (x >> 2); + x = x | (x >> 4); + x = x | (x >> 8); + x = x | (x >> 16); + x = x - ((x >> 1) & 0x55555555); + x = (x & 0x33333333) + ((x >> 2) & 0x33333333); + x = (x + (x >> 4)) & 0x0f0f0f0f; + x = x + (x >> 8); + x = x + (x >> 16); + return (x & 0x0000003f) - 1; +} + +static gint +gst_oss_sink_get_format (GstBufferFormat fmt) +{ + gint result; + + switch (fmt) { + case GST_MU_LAW: + result = AFMT_MU_LAW; + break; + case GST_A_LAW: + result = AFMT_A_LAW; + break; + case GST_IMA_ADPCM: + result = AFMT_IMA_ADPCM; + break; + case GST_U8: + result = AFMT_U8; + break; + case GST_S16_LE: + result = AFMT_S16_LE; + break; + case GST_S16_BE: + result = AFMT_S16_BE; + break; + case GST_S8: + result = AFMT_S8; + break; + case GST_U16_LE: + result = AFMT_U16_LE; + break; + case GST_U16_BE: + result = AFMT_U16_BE; + break; + case GST_MPEG: + result = AFMT_MPEG; + break; + default: + result = 0; + break; + } + return result; +} + +static gboolean +gst_oss_sink_open (GstAudioSink * asink) +{ + GstOssSink *oss; + int mode; + + oss = GST_OSSSINK (asink); + + mode = O_WRONLY; + mode |= O_NONBLOCK; + + oss->fd = open (oss->device, mode, 0); + if (oss->fd == -1) { + switch (errno) { + case EBUSY: + goto busy; + case EACCES: + goto no_permission; + default: + goto open_failed; + } + } + + return TRUE; + + /* ERRORS */ +busy: + { + GST_ELEMENT_ERROR (oss, RESOURCE, BUSY, + (_("Could not open audio device for playback. " + "Device is being used by another application.")), (NULL)); + return FALSE; + } +no_permission: + { + GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE, + (_("Could not open audio device for playback. " + "You don't have permission to open the device.")), + GST_ERROR_SYSTEM); + return FALSE; + } +open_failed: + { + GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE, + (_("Could not open audio device for playback.")), GST_ERROR_SYSTEM); + return FALSE; + } +} + +static gboolean +gst_oss_sink_close (GstAudioSink * asink) +{ + close (GST_OSSSINK (asink)->fd); + GST_OSSSINK (asink)->fd = -1; + return TRUE; +} + +static gboolean +gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec) +{ + GstOssSink *oss; + struct audio_buf_info info; + int mode; + int tmp; + + oss = GST_OSSSINK (asink); + + /* we opened non-blocking so that we can detect if the device is available + * without hanging forever. We now want to remove the non-blocking flag. */ + mode = fcntl (oss->fd, F_GETFL); + mode &= ~O_NONBLOCK; + if (fcntl (oss->fd, F_SETFL, mode) == -1) { + /* some drivers do no support unsetting the non-blocking flag, try to + * close/open the device then. This is racy but we error out properly. */ + gst_oss_sink_close (asink); + if ((oss->fd = open (oss->device, O_WRONLY, 0)) == -1) + goto non_block; + } + + tmp = gst_oss_sink_get_format (spec->format); + if (tmp == 0) + goto wrong_format; + + if (spec->width != 16 && spec->width != 8) + goto dodgy_width; + + SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp, "SETFMT"); + if (spec->channels == 2) + SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO"); + SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS"); + SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED"); + + tmp = ilog2 (spec->segsize); + tmp = ((spec->segtotal & 0x7fff) << 16) | tmp; + GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x", + spec->segsize, spec->segtotal, tmp); + + SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT"); + GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info, "GETOSPACE"); + + spec->segsize = info.fragsize; + spec->segtotal = info.fragstotal; + + spec->bytes_per_sample = (spec->width / 8) * spec->channels; + oss->bytes_per_sample = (spec->width / 8) * spec->channels; + + GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x", + spec->segsize, spec->segtotal, tmp); + + return TRUE; + + /* ERRORS */ +non_block: + { + GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL), + ("Unable to set device %s in non blocking mode: %s", + oss->device, g_strerror (errno))); + return FALSE; + } +wrong_format: + { + GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL), + ("Unable to get format %d", spec->format)); + return FALSE; + } +dodgy_width: + { + GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL), + ("unexpected width %d", spec->width)); + return FALSE; + } +} + +static gboolean +gst_oss_sink_unprepare (GstAudioSink * asink) +{ + /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */ + + if (!gst_oss_sink_close (asink)) + goto couldnt_close; + + if (!gst_oss_sink_open (asink)) + goto couldnt_reopen; + + return TRUE; + + /* ERRORS */ +couldnt_close: + { + GST_DEBUG_OBJECT (asink, "Could not close the audio device"); + return FALSE; + } +couldnt_reopen: + { + GST_DEBUG_OBJECT (asink, "Could not reopen the audio device"); + return FALSE; + } +} + +static guint +gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length) +{ + return write (GST_OSSSINK (asink)->fd, data, length); +} + +static guint +gst_oss_sink_delay (GstAudioSink * asink) +{ + GstOssSink *oss; + gint delay = 0; + gint ret; + + oss = GST_OSSSINK (asink); + +#ifdef SNDCTL_DSP_GETODELAY + ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay); +#else + ret = -1; +#endif + if (ret < 0) { + audio_buf_info info; + + ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info); + + delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes); + } + return delay / oss->bytes_per_sample; +} + +static void +gst_oss_sink_reset (GstAudioSink * asink) +{ + /* There's nothing we can do here really: OSS can't handle access to the + * same device/fd from multiple threads and might deadlock or blow up in + * other ways if we try an ioctl SNDCTL_DSP_RESET or similar */ +} diff --git a/sys/oss/gstosssink.h b/sys/oss/gstosssink.h new file mode 100644 index 0000000..bb2ca93 --- /dev/null +++ b/sys/oss/gstosssink.h @@ -0,0 +1,62 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000 Wim Taymans <wtay@chello.be> + * + * gstosssink.h: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#ifndef __GST_OSSSINK_H__ +#define __GST_OSSSINK_H__ + + +#include <gst/gst.h> +#include <gst/audio/gstaudiosink.h> + +#include "gstosshelper.h" + +G_BEGIN_DECLS + +#define GST_TYPE_OSSSINK (gst_oss_sink_get_type()) +#define GST_OSSSINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSSINK,GstOssSink)) +#define GST_OSSSINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSSINK,GstOssSinkClass)) +#define GST_IS_OSSSINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSSINK)) +#define GST_IS_OSSSINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSSINK)) + +typedef struct _GstOssSink GstOssSink; +typedef struct _GstOssSinkClass GstOssSinkClass; + +struct _GstOssSink { + GstAudioSink sink; + + gchar *device; + gint fd; + gint bytes_per_sample; + + GstCaps *probed_caps; +}; + +struct _GstOssSinkClass { + GstAudioSinkClass parent_class; +}; + +GType gst_oss_sink_get_type(void); + +G_END_DECLS + +#endif /* __GST_OSSSINK_H__ */ diff --git a/sys/oss/gstosssrc.c b/sys/oss/gstosssrc.c new file mode 100644 index 0000000..33f585d --- /dev/null +++ b/sys/oss/gstosssrc.c @@ -0,0 +1,537 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000,2005 Wim Taymans <wim@fluendo.com> + * + * gstosssrc.c: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-osssrc + * + * This element lets you record sound using the Open Sound System (OSS). + * + * <refsect2> + * <title>Example pipelines</title> + * |[ + * gst-launch -v osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg + * ]| will record sound from your sound card using OSS and encode it to an + * Ogg/Vorbis file (this will only work if your mixer settings are right + * and the right inputs enabled etc.) + * </refsect2> + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <sys/ioctl.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <string.h> + +#ifdef HAVE_OSS_INCLUDE_IN_SYS +# include <sys/soundcard.h> +#else +# ifdef HAVE_OSS_INCLUDE_IN_ROOT +# include <soundcard.h> +# else +# ifdef HAVE_OSS_INCLUDE_IN_MACHINE +# include <machine/soundcard.h> +# else +# error "What to include?" +# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */ +# endif /* HAVE_OSS_INCLUDE_IN_ROOT */ +#endif /* HAVE_OSS_INCLUDE_IN_SYS */ + +#include "gstosssrc.h" +#include "common.h" + +#include <gst/gst-i18n-plugin.h> + +GST_DEBUG_CATEGORY_EXTERN (oss_debug); +#define GST_CAT_DEFAULT oss_debug + +#define DEFAULT_DEVICE "/dev/dsp" +#define DEFAULT_DEVICE_NAME "" + +enum +{ + PROP_0, + PROP_DEVICE, + PROP_DEVICE_NAME, +}; + +GST_BOILERPLATE_WITH_INTERFACE (GstOssSrc, gst_oss_src, GstAudioSrc, + GST_TYPE_AUDIO_SRC, GstMixer, GST_TYPE_MIXER, gst_oss_src_mixer); + +GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssSrc, gst_oss_src_mixer); + +static void gst_oss_src_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static void gst_oss_src_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); + +static void gst_oss_src_dispose (GObject * object); +static void gst_oss_src_finalize (GstOssSrc * osssrc); + +static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc); + +static gboolean gst_oss_src_open (GstAudioSrc * asrc); +static gboolean gst_oss_src_close (GstAudioSrc * asrc); +static gboolean gst_oss_src_prepare (GstAudioSrc * asrc, + GstRingBufferSpec * spec); +static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc); +static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length); +static guint gst_oss_src_delay (GstAudioSrc * asrc); +static void gst_oss_src_reset (GstAudioSrc * asrc); + + + +static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-raw-int, " + "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, " + "signed = (boolean) { TRUE, FALSE }, " + "width = (int) 16, " + "depth = (int) 16, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; " + "audio/x-raw-int, " + "signed = (boolean) { TRUE, FALSE }, " + "width = (int) 8, " + "depth = (int) 8, " + "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]") + ); + + +static void +gst_oss_src_dispose (GObject * object) +{ + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_oss_src_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_set_details_simple (element_class, "Audio Source (OSS)", + "Source/Audio", + "Capture from a sound card via OSS", + "Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>"); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&osssrc_src_factory)); +} + +static void +gst_oss_src_class_init (GstOssSrcClass * klass) +{ + GObjectClass *gobject_class; + GstBaseSrcClass *gstbasesrc_class; + GstAudioSrcClass *gstaudiosrc_class; + + gobject_class = (GObjectClass *) klass; + gstbasesrc_class = (GstBaseSrcClass *) klass; + gstaudiosrc_class = (GstAudioSrcClass *) klass; + + gobject_class->dispose = gst_oss_src_dispose; + gobject_class->finalize = (GObjectFinalizeFunc) gst_oss_src_finalize; + gobject_class->get_property = gst_oss_src_get_property; + gobject_class->set_property = gst_oss_src_set_property; + + gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_src_getcaps); + + gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss_src_open); + gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_src_prepare); + gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_src_unprepare); + gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss_src_close); + gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss_src_read); + gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss_src_delay); + gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss_src_reset); + + g_object_class_install_property (gobject_class, PROP_DEVICE, + g_param_spec_string ("device", "Device", + "OSS device (usually /dev/dspN)", DEFAULT_DEVICE, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_DEVICE_NAME, + g_param_spec_string ("device-name", "Device name", + "Human-readable name of the sound device", DEFAULT_DEVICE_NAME, + G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); +} + +static void +gst_oss_src_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstOssSrc *src; + + src = GST_OSS_SRC (object); + + switch (prop_id) { + case PROP_DEVICE: + if (src->device) + g_free (src->device); + src->device = g_value_dup_string (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_oss_src_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstOssSrc *src; + + src = GST_OSS_SRC (object); + + switch (prop_id) { + case PROP_DEVICE: + g_value_set_string (value, src->device); + break; + case PROP_DEVICE_NAME: + g_value_set_string (value, src->device_name); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_oss_src_init (GstOssSrc * osssrc, GstOssSrcClass * g_class) +{ + const gchar *device; + + GST_DEBUG ("initializing osssrc"); + + device = g_getenv ("AUDIODEV"); + if (device == NULL) + device = DEFAULT_DEVICE; + + osssrc->fd = -1; + osssrc->device = g_strdup (device); + osssrc->device_name = g_strdup (DEFAULT_DEVICE_NAME); + osssrc->probed_caps = NULL; +} + +static void +gst_oss_src_finalize (GstOssSrc * osssrc) +{ + g_free (osssrc->device); + g_free (osssrc->device_name); + + G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (osssrc)); +} + +static GstCaps * +gst_oss_src_getcaps (GstBaseSrc * bsrc) +{ + GstOssSrc *osssrc; + GstCaps *caps; + + osssrc = GST_OSS_SRC (bsrc); + + if (osssrc->fd == -1) { + GST_DEBUG_OBJECT (osssrc, "device not open, using template caps"); + return NULL; /* base class will get template caps for us */ + } + + if (osssrc->probed_caps) { + GST_LOG_OBJECT (osssrc, "Returning cached caps"); + return gst_caps_ref (osssrc->probed_caps); + } + + caps = gst_oss_helper_probe_caps (osssrc->fd); + + if (caps) { + osssrc->probed_caps = gst_caps_ref (caps); + } + + GST_INFO_OBJECT (osssrc, "returning caps %" GST_PTR_FORMAT, caps); + + return caps; +} + +static gint +ilog2 (gint x) +{ + /* well... hacker's delight explains... */ + x = x | (x >> 1); + x = x | (x >> 2); + x = x | (x >> 4); + x = x | (x >> 8); + x = x | (x >> 16); + x = x - ((x >> 1) & 0x55555555); + x = (x & 0x33333333) + ((x >> 2) & 0x33333333); + x = (x + (x >> 4)) & 0x0f0f0f0f; + x = x + (x >> 8); + x = x + (x >> 16); + return (x & 0x0000003f) - 1; +} + +static gint +gst_oss_src_get_format (GstBufferFormat fmt) +{ + gint result; + + switch (fmt) { + case GST_MU_LAW: + result = AFMT_MU_LAW; + break; + case GST_A_LAW: + result = AFMT_A_LAW; + break; + case GST_IMA_ADPCM: + result = AFMT_IMA_ADPCM; + break; + case GST_U8: + result = AFMT_U8; + break; + case GST_S16_LE: + result = AFMT_S16_LE; + break; + case GST_S16_BE: + result = AFMT_S16_BE; + break; + case GST_S8: + result = AFMT_S8; + break; + case GST_U16_LE: + result = AFMT_U16_LE; + break; + case GST_U16_BE: + result = AFMT_U16_BE; + break; + case GST_MPEG: + result = AFMT_MPEG; + break; + default: + result = 0; + break; + } + return result; +} + +static gboolean +gst_oss_src_open (GstAudioSrc * asrc) +{ + GstOssSrc *oss; + int mode; + + oss = GST_OSS_SRC (asrc); + + mode = O_RDONLY; + mode |= O_NONBLOCK; + + oss->fd = open (oss->device, mode, 0); + if (oss->fd == -1) { + switch (errno) { + case EACCES: + goto no_permission; + default: + goto open_failed; + } + } + + if (!oss->mixer) { + oss->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_CAPTURE); + + if (oss->mixer) { + g_free (oss->device_name); + oss->device_name = g_strdup (oss->mixer->cardname); + } + } + return TRUE; + +no_permission: + { + GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, + (_("Could not open audio device for recording. " + "You don't have permission to open the device.")), + GST_ERROR_SYSTEM); + return FALSE; + } +open_failed: + { + GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, + (_("Could not open audio device for recording.")), + ("Unable to open device %s for recording: %s", + oss->device, g_strerror (errno))); + return FALSE; + } +} + +static gboolean +gst_oss_src_close (GstAudioSrc * asrc) +{ + GstOssSrc *oss; + + oss = GST_OSS_SRC (asrc); + + close (oss->fd); + + if (oss->mixer) { + gst_ossmixer_free (oss->mixer); + oss->mixer = NULL; + } + + gst_caps_replace (&oss->probed_caps, NULL); + + return TRUE; +} + +static gboolean +gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec) +{ + GstOssSrc *oss; + struct audio_buf_info info; + int mode; + int fmt, tmp; + + oss = GST_OSS_SRC (asrc); + + mode = fcntl (oss->fd, F_GETFL); + mode &= ~O_NONBLOCK; + if (fcntl (oss->fd, F_SETFL, mode) == -1) + goto non_block; + + fmt = gst_oss_src_get_format (spec->format); + if (fmt == 0) + goto wrong_format; + + tmp = ilog2 (spec->segsize); + tmp = ((spec->segtotal & 0x7fff) << 16) | tmp; + GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x", + spec->segsize, spec->segtotal, tmp); + + SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT"); + + SET_PARAM (oss, SNDCTL_DSP_RESET, 0, "RESET"); + + SET_PARAM (oss, SNDCTL_DSP_SETFMT, fmt, "SETFMT"); + if (spec->channels == 2) + SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO"); + SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS"); + SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED"); + + GET_PARAM (oss, SNDCTL_DSP_GETISPACE, &info, "GETISPACE"); + + spec->segsize = info.fragsize; + spec->segtotal = info.fragstotal; + + if (spec->width != 16 && spec->width != 8) + goto dodgy_width; + + spec->bytes_per_sample = (spec->width / 8) * spec->channels; + oss->bytes_per_sample = (spec->width / 8) * spec->channels; + memset (spec->silence_sample, 0, spec->bytes_per_sample); + + GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x", + spec->segsize, spec->segtotal, tmp); + + return TRUE; + +non_block: + { + GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, + ("Unable to set device %s in non blocking mode: %s", + oss->device, g_strerror (errno)), (NULL)); + return FALSE; + } +wrong_format: + { + GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, + ("Unable to get format %d", spec->format), (NULL)); + return FALSE; + } +dodgy_width: + { + GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, + ("Unexpected width %d", spec->width), (NULL)); + return FALSE; + } +} + +static gboolean +gst_oss_src_unprepare (GstAudioSrc * asrc) +{ + /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */ + + if (!gst_oss_src_close (asrc)) + goto couldnt_close; + + if (!gst_oss_src_open (asrc)) + goto couldnt_reopen; + + return TRUE; + +couldnt_close: + { + GST_DEBUG_OBJECT (asrc, "Could not close the audio device"); + return FALSE; + } +couldnt_reopen: + { + GST_DEBUG_OBJECT (asrc, "Could not reopen the audio device"); + return FALSE; + } +} + +static guint +gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length) +{ + return read (GST_OSS_SRC (asrc)->fd, data, length); +} + +static guint +gst_oss_src_delay (GstAudioSrc * asrc) +{ + GstOssSrc *oss; + gint delay = 0; + gint ret; + + oss = GST_OSS_SRC (asrc); + +#ifdef SNDCTL_DSP_GETODELAY + ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay); +#else + ret = -1; +#endif + if (ret < 0) { + audio_buf_info info; + + ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info); + + delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes); + } + return delay / oss->bytes_per_sample; +} + +static void +gst_oss_src_reset (GstAudioSrc * asrc) +{ + /* There's nothing we can do here really: OSS can't handle access to the + * same device/fd from multiple threads and might deadlock or blow up in + * other ways if we try an ioctl SNDCTL_DSP_RESET or similar */ +} diff --git a/sys/oss/gstosssrc.h b/sys/oss/gstosssrc.h new file mode 100644 index 0000000..0d14613 --- /dev/null +++ b/sys/oss/gstosssrc.h @@ -0,0 +1,67 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000 Wim Taymans <wtay@chello.be> + * + * gstosssrc.h: + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#ifndef __GST_OSS_SRC_H__ +#define __GST_OSS_SRC_H__ + + +#include <gst/gst.h> +#include <gst/audio/gstaudiosrc.h> + +#include "gstosshelper.h" +#include "gstossmixer.h" + +G_BEGIN_DECLS + +#define GST_TYPE_OSS_SRC (gst_oss_src_get_type()) +#define GST_OSS_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS_SRC,GstOssSrc)) +#define GST_OSS_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS_SRC,GstOssSrcClass)) +#define GST_IS_OSS_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS_SRC)) +#define GST_IS_OSS_SRC_CLASS(klas) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS_SRC)) + +typedef struct _GstOssSrc GstOssSrc; +typedef struct _GstOssSrcClass GstOssSrcClass; + +struct _GstOssSrc { + GstAudioSrc src; + + gint fd; + gint bytes_per_sample; + + gchar *device; + gchar *device_name; + + GstCaps *probed_caps; + + GstOssMixer *mixer; +}; + +struct _GstOssSrcClass { + GstAudioSrcClass parent_class; +}; + +GType gst_oss_src_get_type(void); + +G_END_DECLS + +#endif /* __GST_OSS_SRC_H__ */ |