diff options
author | Thomas Vander Stichele <thomas@apestaart.org> | 2001-12-17 19:03:13 +0000 |
---|---|---|
committer | Thomas Vander Stichele <thomas@apestaart.org> | 2001-12-17 19:03:13 +0000 |
commit | f0bb7eef82576f581bdd205f2c9eb4278223fcc7 (patch) | |
tree | 530e9dea1ad3117141c1682318e74811d1fcb86a | |
parent | 754313ab71cd4cb046191b3459842a063fd3642b (diff) |
first batch
Original commit message from CVS:
first batch
-rw-r--r-- | sys/Makefile.am | 5 | ||||
-rw-r--r-- | sys/oss/Makefile.am | 11 | ||||
-rw-r--r-- | sys/oss/README | 37 | ||||
-rw-r--r-- | sys/oss/gstossaudio.c | 47 | ||||
-rw-r--r-- | sys/oss/gstossgst.c | 448 | ||||
-rw-r--r-- | sys/oss/gstossgst.h | 87 | ||||
-rw-r--r-- | sys/oss/gstosshelper.c | 401 | ||||
-rw-r--r-- | sys/oss/gstosshelper.h | 44 | ||||
-rw-r--r-- | sys/oss/gstosssink.c | 626 | ||||
-rw-r--r-- | sys/oss/gstosssink.h | 96 | ||||
-rw-r--r-- | sys/oss/gstosssrc.c | 419 | ||||
-rw-r--r-- | sys/oss/gstosssrc.h | 94 |
12 files changed, 2315 insertions, 0 deletions
diff --git a/sys/Makefile.am b/sys/Makefile.am new file mode 100644 index 00000000..101ea5a5 --- /dev/null +++ b/sys/Makefile.am @@ -0,0 +1,5 @@ +### use HAVE_ stuff to decide on dirs +DIRS=qcam v4l vcdsrc vgasink xvideosink + +DIST_SUBDIRS=qcam v4l vcdsrc vgasink xvideosink + diff --git a/sys/oss/Makefile.am b/sys/oss/Makefile.am new file mode 100644 index 00000000..c6c84fd6 --- /dev/null +++ b/sys/oss/Makefile.am @@ -0,0 +1,11 @@ +filterdir = $(libdir)/gst + +filter_LTLIBRARIES = libgstossaudio.la libgstosshelper.la + +libgstossaudio_la_SOURCES = gstosssink.c gstosssrc.c gstossaudio.c gstossgst.c +libgstossaudio_la_CFLAGS = $(GST_CFLAGS) + +libgstosshelper_la_SOURCES = gstosshelper.c + +noinst_HEADERS = gstosssink.h gstosssrc.h gstossgst.h gstosshelper.h + diff --git a/sys/oss/README b/sys/oss/README new file mode 100644 index 00000000..db3f1db9 --- /dev/null +++ b/sys/oss/README @@ -0,0 +1,37 @@ + + + GStreamer + + + (------------------------------------) + ! ! + sink GstOss src + ! ! + ! ! + (------------------------------------) + ! ^ + ! 500 ! 501 + V ! + (------------------------------------) + ! GstOssHelper ! + (------------------------------------) + ! Native OSS APP ! + ! ! + (------------------------------------) + + +Port 500 protocol +----------------- + +nothing yet + +port 501 protocol +----------------- + +1 <N> <N-bytes> + N bytes of raw audio data following + +2 <OSS format> + + + diff --git a/sys/oss/gstossaudio.c b/sys/oss/gstossaudio.c new file mode 100644 index 00000000..5ffd6248 --- /dev/null +++ b/sys/oss/gstossaudio.c @@ -0,0 +1,47 @@ +/* Gnome-Streamer + * 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. + */ + + +#include "gstosssink.h" +#include "gstosssrc.h" +#include "gstossgst.h" + +static gboolean +plugin_init (GModule *module, GstPlugin *plugin) +{ + gboolean ret; + + ret = gst_osssink_factory_init (plugin); + g_return_val_if_fail (ret == TRUE, FALSE); + + ret = gst_osssrc_factory_init (plugin); + g_return_val_if_fail (ret == TRUE, FALSE); + + ret = gst_ossgst_factory_init (plugin); + g_return_val_if_fail (ret == TRUE, FALSE); + + return TRUE; +} + +GstPluginDesc plugin_desc = { + GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "ossaudio", + plugin_init +}; diff --git a/sys/oss/gstossgst.c b/sys/oss/gstossgst.c new file mode 100644 index 00000000..332f8214 --- /dev/null +++ b/sys/oss/gstossgst.c @@ -0,0 +1,448 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000 Wim Taymans <wim.taymans@chello.be> + * + * gstossgst.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. + */ + + +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <fcntl.h> +#include <sys/soundcard.h> +#include <unistd.h> +#include <errno.h> +#include <stdlib.h> + +#include "gstossgst.h" + +#include "gstosshelper.h" + +static GstElementDetails gst_ossgst_details = { + "Audio Wrapper (OSS)", + "Src/Audio", + "Hijacks /dev/dsp to get the output of OSS apps into GStreamer", + VERSION, + "Wim Taymans <wim.taymans@chello.be>", + "(C) 2001", +}; + +static void gst_ossgst_class_init (GstOssGstClass *klass); +static void gst_ossgst_init (GstOssGst *ossgst); + +static GstElementStateReturn gst_ossgst_change_state (GstElement *element); + +static void gst_ossgst_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec); +static void gst_ossgst_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec); + +static GstBuffer* gst_ossgst_get (GstPad *pad); + +/* OssGst signals and args */ +enum { + LAST_SIGNAL +}; + +enum { + ARG_0, + ARG_MUTE, + ARG_PROGRAM, + /* FILL ME */ +}; + +static GstPadTemplate* +ossgst_src_factory (void) +{ + return + gst_padtemplate_new ( + "src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + gst_caps_new ( + "ossgst_src", + "audio/raw", + gst_props_new ( + "format", GST_PROPS_STRING ("int"), + "law", GST_PROPS_INT (0), + "endianness", GST_PROPS_INT (G_BYTE_ORDER), + "signed", GST_PROPS_LIST ( + GST_PROPS_BOOLEAN (FALSE), + GST_PROPS_BOOLEAN (TRUE) + ), + "width", GST_PROPS_LIST ( + GST_PROPS_INT (8), + GST_PROPS_INT (16) + ), + "depth", GST_PROPS_LIST ( + GST_PROPS_INT (8), + GST_PROPS_INT (16) + ), + "rate", GST_PROPS_INT_RANGE (8000, 48000), + "channels", GST_PROPS_INT_RANGE (1, 2), + NULL)), + NULL); +} + + +static GstElementClass *parent_class = NULL; +static GstPadTemplate *gst_ossgst_src_template; + +static gchar *plugin_dir = NULL; + +GType +gst_ossgst_get_type (void) +{ + static GType ossgst_type = 0; + + if (!ossgst_type) { + static const GTypeInfo ossgst_info = { + sizeof(GstOssGstClass), + NULL, + NULL, + (GClassInitFunc)gst_ossgst_class_init, + NULL, + NULL, + sizeof(GstOssGst), + 0, + (GInstanceInitFunc)gst_ossgst_init, + }; + ossgst_type = g_type_register_static (GST_TYPE_ELEMENT, "GstOssGst", &ossgst_info, 0); + } + + return ossgst_type; +} + +static void +gst_ossgst_class_init (GstOssGstClass *klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass*)klass; + gstelement_class = (GstElementClass*)klass; + + parent_class = g_type_class_ref(GST_TYPE_ELEMENT); + + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_MUTE, + g_param_spec_boolean("mute","mute","mute", + TRUE,G_PARAM_READWRITE)); // CHECKME + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_PROGRAM, + g_param_spec_string("command","command","command", + NULL, G_PARAM_READWRITE)); // CHECKME + + gobject_class->set_property = gst_ossgst_set_property; + gobject_class->get_property = gst_ossgst_get_property; + + gstelement_class->change_state = gst_ossgst_change_state; +} + +static void +gst_ossgst_init (GstOssGst *ossgst) +{ + ossgst->srcpad = gst_pad_new_from_template (gst_ossgst_src_template, "src"); + gst_element_add_pad (GST_ELEMENT (ossgst), ossgst->srcpad); + + gst_pad_set_get_function (ossgst->srcpad, gst_ossgst_get); + + ossgst->command = NULL; +} + +static GstCaps* +gst_ossgst_format_to_caps (gint format, gint stereo, gint rate) +{ + GstCaps *caps = NULL; + gint law = 0; + gulong endianness = G_BYTE_ORDER; + gboolean is_signed = TRUE; + gint width = 16; + gboolean supported = TRUE; + + GST_DEBUG (0, "have format 0x%08x %d %d\n", format, stereo, rate); + + switch (format) { + case AFMT_MU_LAW: + law = 1; + break; + case AFMT_A_LAW: + law = 2; + break; + case AFMT_U8: + width = 8; + is_signed = FALSE; + break; + case AFMT_S16_LE: + width = 16; + endianness = G_LITTLE_ENDIAN; + is_signed = TRUE; + break; + case AFMT_S16_BE: + endianness = G_BIG_ENDIAN; + width = 16; + is_signed = TRUE; + break; + case AFMT_S8: + width = 8; + is_signed = TRUE; + break; + case AFMT_U16_LE: + width = 16; + endianness = G_LITTLE_ENDIAN; + is_signed = FALSE; + break; + case AFMT_U16_BE: + width = 16; + endianness = G_BIG_ENDIAN; + is_signed = FALSE; + break; + case AFMT_IMA_ADPCM: + case AFMT_MPEG: +#ifdef AFMT_AC3 + case AFMT_AC3: +#endif + default: + supported = FALSE; + break; + } + + if (supported) { + caps = gst_caps_new ( + "ossgst_caps", + "audio/raw", + gst_props_new ( + "format", GST_PROPS_STRING ("int"), + "law", GST_PROPS_INT (law), + "endianness", GST_PROPS_INT (endianness), + "signed", GST_PROPS_BOOLEAN (is_signed), + "width", GST_PROPS_INT (width), + "depth", GST_PROPS_INT (width), + "rate", GST_PROPS_INT (rate), + "channels", GST_PROPS_INT (stereo?2:1), + NULL)); + } + else { + g_warning ("gstossgst: program tried to use unsupported format %x\n", format); + } + + return caps; +} + +static GstBuffer* +gst_ossgst_get (GstPad *pad) +{ + GstOssGst *ossgst; + GstBuffer *buf = NULL; + command cmd; + gboolean have_data = FALSE; + + g_return_val_if_fail (pad != NULL, NULL); + g_return_val_if_fail (GST_IS_PAD (pad), NULL); + + /* this has to be an audio buffer */ + ossgst = GST_OSSGST (gst_pad_get_parent (pad)); + + while (!have_data) { + /* read the command */ + read (ossgst->fdout[0], &cmd, sizeof (command)); + + switch (cmd.id) { + case CMD_DATA: + buf = gst_buffer_new (); + GST_BUFFER_SIZE (buf) = cmd.cmd.length; + GST_BUFFER_DATA (buf) = g_malloc (GST_BUFFER_SIZE (buf)); + + GST_BUFFER_SIZE (buf) = read (ossgst->fdout[0], GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); + have_data = TRUE; + break; + case CMD_FORMAT: + { + GstCaps *caps; + + caps = gst_ossgst_format_to_caps (cmd.cmd.format.format, + cmd.cmd.format.stereo, + cmd.cmd.format.rate); + + gst_pad_set_caps (ossgst->srcpad, caps); + } + break; + default: + break; + } + } + + return buf; +} + +static void +gst_ossgst_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) +{ + GstOssGst *ossgst; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_OSSGST (object)); + + ossgst = GST_OSSGST (object); + + switch (prop_id) { + case ARG_MUTE: + ossgst->mute = g_value_get_boolean (value); + break; + case ARG_PROGRAM: + if (ossgst->command) + g_free (ossgst->command); + ossgst->command = g_strdup (g_value_get_string (value)); + break; + default: + break; + } +} + +static void +gst_ossgst_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) +{ + GstOssGst *ossgst; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_OSSGST (object)); + + ossgst = GST_OSSGST (object); + + switch (prop_id) { + case ARG_MUTE: + g_value_set_boolean (value, ossgst->mute); + break; + case ARG_PROGRAM: + g_value_set_string (value, ossgst->command); + break; + default: + break; + } +} + +static gboolean +gst_ossgst_spawn_process (GstOssGst *ossgst) +{ + static gchar *ld_preload; + + pipe(ossgst->fdin); + pipe(ossgst->fdout); + + GST_DEBUG (0, "about to fork\n"); + + if((ossgst->childpid = fork()) == -1) + { + perror("fork"); + gst_element_error(GST_ELEMENT(ossgst),"forking"); + return FALSE; + } + GST_DEBUG (0,"forked %d\n", ossgst->childpid); + + if(ossgst->childpid == 0) + { + gchar **args; + + GST_DEBUG (0, "fork command %d\n", ossgst->childpid); + + ld_preload = getenv ("LD_PRELOAD"); + + if (ld_preload == NULL) { + ld_preload = ""; + } + + ld_preload = g_strconcat (ld_preload, " ", plugin_dir, G_DIR_SEPARATOR_S, + "libgstosshelper.so", NULL); + + setenv ("LD_PRELOAD", ld_preload, TRUE); + + // child + dup2(ossgst->fdin[0], HELPER_MAGIC_IN); /* set the childs input stream */ + dup2(ossgst->fdout[1], HELPER_MAGIC_OUT); /* set the childs output stream */ + + // split the arguments + args = g_strsplit (ossgst->command, " ", 0); + + execvp(args[0], args); + + // will only reach if error + perror("exec"); + gst_element_error(GST_ELEMENT(ossgst),"starting child process"); + return FALSE; + + } + GST_FLAG_SET(ossgst,GST_OSSGST_OPEN); + + return TRUE; +} + +static gboolean +gst_ossgst_kill_process (GstOssGst *ossgst) +{ + return TRUE; +} + +static GstElementStateReturn +gst_ossgst_change_state (GstElement *element) +{ + g_return_val_if_fail (GST_IS_OSSGST (element), FALSE); + + if (GST_STATE_PENDING (element) == GST_STATE_NULL) { + if (GST_FLAG_IS_SET (element, GST_OSSGST_OPEN)) + gst_ossgst_kill_process (GST_OSSGST (element)); + } else { + if (!GST_FLAG_IS_SET (element, GST_OSSGST_OPEN)) { + if (!gst_ossgst_spawn_process (GST_OSSGST (element))) { + return GST_STATE_FAILURE; + } + } + } + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + return GST_STATE_SUCCESS; +} + +gboolean +gst_ossgst_factory_init (GstPlugin *plugin) +{ + GstElementFactory *factory; + gchar **path; + gint i =0; + + // get the path of this plugin, we assume the helper progam lives in the + // same directory. + path = g_strsplit (plugin->filename, G_DIR_SEPARATOR_S, 0); + while (path[i]) { + i++; + if (path[i] == NULL) { + g_free (path[i-1]); + path[i-1] = NULL; + } + } + plugin_dir = g_strjoinv (G_DIR_SEPARATOR_S, path); + g_strfreev (path); + + factory = gst_elementfactory_new ("ossgst", GST_TYPE_OSSGST, &gst_ossgst_details); + g_return_val_if_fail (factory != NULL, FALSE); + + gst_ossgst_src_template = ossgst_src_factory (); + gst_elementfactory_add_padtemplate (factory, gst_ossgst_src_template); + + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); + + return TRUE; +} + diff --git a/sys/oss/gstossgst.h b/sys/oss/gstossgst.h new file mode 100644 index 00000000..411f771e --- /dev/null +++ b/sys/oss/gstossgst.h @@ -0,0 +1,87 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000 Wim Taymans <wtay@chello.be> + * + * gstossgst.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_OSSGST_H__ +#define __GST_OSSGST_H__ + + +#include <config.h> +#include <gst/gst.h> + +#include <sys/types.h> + +#ifdef __cplusplus +extern "C" { +#endif /* __cplusplus */ + + +#define GST_TYPE_OSSGST \ + (gst_ossgst_get_type()) +#define GST_OSSGST(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSGST,GstOssGst)) +#define GST_OSSGST_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSGST,GstOssGstClass)) +#define GST_IS_OSSGST(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSGST)) +#define GST_IS_OSSGST_CLASS(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSGST)) + +typedef enum { + GST_OSSGST_OPEN = GST_ELEMENT_FLAG_LAST, + + GST_OSSGST_FLAG_LAST = GST_ELEMENT_FLAG_LAST+2, +} GstOssGstFlags; + +typedef struct _GstOssGst GstOssGst; +typedef struct _GstOssGstClass GstOssGstClass; + +struct _GstOssGst { + GstElement element; + + GstPad *srcpad; + + gint fdout[2]; + gint fdin[2]; + pid_t childpid; + + /* soundcard state */ + gboolean mute; + gchar *command; +}; + +struct _GstOssGstClass { + GstElementClass parent_class; + + /* signals */ +}; + +GType gst_ossgst_get_type(void); + +gboolean gst_ossgst_factory_init(GstPlugin *plugin); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + + +#endif /* __GST_OSSGST_H__ */ diff --git a/sys/oss/gstosshelper.c b/sys/oss/gstosshelper.c new file mode 100644 index 00000000..a47840a5 --- /dev/null +++ b/sys/oss/gstosshelper.c @@ -0,0 +1,401 @@ +/* Evil evil evil hack to get OSS apps to cooperate with esd + * Copyright (C) 1998, 1999 Manish Singh <yosh@gimp.org> + * + * 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 DSP_DEBUG + +/* This lets you run multiple instances of x11amp by setting the X11AMPNUM + environment variable. Only works on glibc2. + */ +/* #define MULTIPLE_X11AMP */ + +#if defined(__GNUC__) && !defined(__STRICT_ANSI__) + +#ifdef DSP_DEBUG +#define DPRINTF(format, args...) printf(format, ## args) +#else +#define DPRINTF(format, args...) +#endif + + +#include "config.h" + +#include <dlfcn.h> +#include <stdarg.h> +#include <stdlib.h> +#include <string.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <stdio.h> + +#include <errno.h> + +#ifdef HAVE_MACHINE_SOUNDCARD_H +# include <machine/soundcard.h> +#else +# ifdef HAVE_SOUNDCARD_H +# include <soundcard.h> +# else +# include <sys/soundcard.h> +# endif +#endif + +#include "gstosshelper.h" + +/* BSDI has this functionality, but not define :() */ +#if defined(RTLD_NEXT) +#define REAL_LIBC RTLD_NEXT +#else +#define REAL_LIBC ((void *) -1L) +#endif + +#if defined(__FreeBSD__) || defined(__bsdi__) +typedef unsigned long request_t; +#else +typedef int request_t; +#endif + +static int sndfd = -1; +static int new_format = 1; +static int fmt = AFMT_S16_LE; +static int speed = 44100; +static int stereo = 1; + +int +open (const char *pathname, int flags, ...) +{ + static int (*func) (const char *, int, mode_t) = NULL; + va_list args; + mode_t mode; + + if (!func) + func = (int (*) (const char *, int, mode_t)) dlsym (REAL_LIBC, "open"); + + va_start (args, flags); + mode = va_arg (args, mode_t); + va_end (args); + + if (!strcmp (pathname, "/dev/dsp")) { + DPRINTF ("hijacking /dev/dsp open, and taking it to GStreamer...\n"); + return (sndfd = HELPER_MAGIC_SNDFD); + } + return (sndfd = (*func) (pathname, flags, mode)); +} + +static int +dspctl (int fd, request_t request, void *argp) +{ + int *arg = (int *) argp; + + DPRINTF ("hijacking /dev/dsp ioctl, and sending it to GStreamer " + "(%d : %x - %p)\n", fd, request, argp); + + switch (request) + { + case SNDCTL_DSP_RESET: + case SNDCTL_DSP_POST: + break; + + case SNDCTL_DSP_SETFMT: + fmt = *arg; + new_format = 1; + break; + + case SNDCTL_DSP_SPEED: + speed = *arg; + new_format = 1; + break; + + case SNDCTL_DSP_STEREO: + stereo = *arg; + new_format = 1; + break; + + case SNDCTL_DSP_GETBLKSIZE: + *arg = 4096; + break; + + case SNDCTL_DSP_GETFMTS: + *arg = 0x38; + break; + +#ifdef SNDCTL_DSP_GETCAPS + case SNDCTL_DSP_GETCAPS: + *arg = 0; + break; +#endif + + case SNDCTL_DSP_GETOSPACE: + { + audio_buf_info *bufinfo = (audio_buf_info *) argp; + bufinfo->bytes = 4096; + } + break; + + + default: + DPRINTF ("unhandled /dev/dsp ioctl (%x - %p)\n", request, argp); + break; + } + + return 0; +} + +void * +mmap(void *start, size_t length, int prot , int flags, int fd, off_t offset) +{ + static void * (*func) (void *, size_t, int, int, int, off_t) = NULL; + + if (!func) + func = (void * (*) (void *, size_t, int, int, int, off_t)) dlsym (REAL_LIBC, "mmap"); + + if ((fd == sndfd) && (sndfd != -1)) + { + DPRINTF("MMAP: oops... we're in trouble here. /dev/dsp mmap()ed. Not supported yet.\n"); + errno = EACCES; + return (void *)-1; /* Better causing an error than silently not working, in this case */ + } + + return (*func) (start, length, prot, flags, fd, offset); +} + +ssize_t +write (int fd, const void *buf, size_t len) +{ + static int (*func) (int, const void *, size_t) = NULL; + command cmd; + + if (!func) + func = (int (*) (int, const void *, size_t)) dlsym (REAL_LIBC, "write"); + + if ((fd != sndfd) || (sndfd == -1)) + { + return (*func) (fd, buf, len); + } + + DPRINTF("WRITE: called for %d bytes\n", len); + + if (new_format) { + new_format = 0; + + cmd.id = CMD_FORMAT; + cmd.cmd.format.format = fmt; + cmd.cmd.format.stereo = stereo; + cmd.cmd.format.rate = speed; + + (*func) (HELPER_MAGIC_OUT, &cmd, sizeof(command)); + } + cmd.id = CMD_DATA; + cmd.cmd.length = len; + + (*func) (HELPER_MAGIC_OUT, &cmd, sizeof(command)); + (*func) (HELPER_MAGIC_OUT, buf, len); + + //return (*func) (fd, buf, len); + + return len; +} + +int +select (int n, fd_set *readfds, fd_set *writefds, + fd_set *exceptfds, struct timeval *timeout) +{ + static int (*func) (int, fd_set *, fd_set *, fd_set *, struct timeval *) = NULL; + + if (!func) + func = (int (*) (int, fd_set *, fd_set *, fd_set *, struct timeval *)) dlsym (REAL_LIBC, "select"); + + if (n == sndfd) { + DPRINTF ("audiooss: hijacking /dev/dsp select() [output]\n"); + } + + return (*func) (n, readfds, writefds, exceptfds, timeout); +} + +int +dup2 (int oldfd, int newfd) +{ + static int (*func) (int, int) = NULL; + + if (!func) + func = (int (*) (int, int)) dlsym (REAL_LIBC, "dup2"); + + if ((oldfd == sndfd) && (oldfd != -1) && (newfd != -1)) + { + DPRINTF("dup2(%d,%d) (oldfd == sndfd) called\n", oldfd, newfd); + + /* Do not close(newfd) as that would mark it available for reuse by the system - + * just tell the program that yes, we got the fd you asked for. Hackish. */ + sndfd = newfd; + return newfd; + } + return (*func) (oldfd, newfd); +} + +int +ioctl (int fd, request_t request, ...) +{ + static int (*func) (int, request_t, void *) = NULL; + va_list args; + void *argp; + + if (!func) + func = (int (*) (int, request_t, void *)) dlsym (REAL_LIBC, "ioctl"); + + va_start (args, request); + argp = va_arg (args, void *); + va_end (args); + + if (fd == sndfd) + return dspctl (fd, request, argp); + + return (*func) (fd, request, argp); +} + +int +fcntl(int fd, int cmd, ...) +{ + static int (*func) (int, int, void *) = NULL; + va_list args; + void *argp; + + if (!func) + func = (int (*) (int, int, void *)) dlsym (REAL_LIBC, "fcntl"); + + va_start (args, cmd); + argp = va_arg (args, void *); + va_end (args); + + if ((fd != -1) && (fd == sndfd)) + { + DPRINTF ("hijacking /dev/dsp fcntl() " + "(%d : %x - %p)\n", fd, cmd, argp); + if (cmd == F_GETFL) return O_RDWR; + if (cmd == F_GETFD) return sndfd; + return 0; + } + else + { + return (*func) (fd, cmd, argp); + } + return 0; +} + +int +close (int fd) +{ + static int (*func) (int) = NULL; + + if (!func) + func = (int (*) (int)) dlsym (REAL_LIBC, "close"); + + if (fd == sndfd) + sndfd = -1; + + return (*func) (fd); +} + +#ifdef MULTIPLE_X11AMP + +#include <socketbits.h> +#include <sys/param.h> +#include <sys/un.h> + +#define ENVSET "X11AMPNUM" + +int +unlink (const char *filename) +{ + static int (*func) (const char *) = NULL; + char *num; + + if (!func) + func = (int (*) (const char *)) dlsym (REAL_LIBC, "unlink"); + + if (!strcmp (filename, "/tmp/X11Amp_CTRL") && (num = getenv (ENVSET))) + { + char buf[PATH_MAX] = "/tmp/X11Amp_CTRL"; + strcat (buf, num); + return (*func) (buf); + } + else + return (*func) (filename); +} + +typedef int (*sa_func_t) (int, struct sockaddr *, int); + +static int +sockaddr_mangle (sa_func_t func, int fd, struct sockaddr *addr, int len) +{ + char *num; + + if (!strcmp (((struct sockaddr_un *) addr)->sun_path, "/tmp/X11Amp_CTRL") + && (num = getenv(ENVSET))) + { + int ret; + char buf[PATH_MAX] = "/tmp/X11Amp_CTRL"; + + struct sockaddr_un *new_addr = malloc (len); + + strcat (buf, num); + memcpy (new_addr, addr, len); + strcpy (new_addr->sun_path, buf); + + ret = (*func) (fd, (struct sockaddr *) new_addr, len); + + free (new_addr); + return ret; + } + else + return (*func) (fd, addr, len); +} + +int +bind (int fd, struct sockaddr *addr, int len) +{ + static sa_func_t func = NULL; + + if (!func) + func = (sa_func_t) dlsym (REAL_LIBC, "bind"); + return sockaddr_mangle (func, fd, addr, len); +} + +int +connect (int fd, struct sockaddr *addr, int len) +{ + static sa_func_t func = NULL; + + if (!func) + func = (sa_func_t) dlsym (REAL_LIBC, "connect"); + return sockaddr_mangle (func, fd, addr, len); +} + +#endif /* MULTIPLE_X11AMP */ + +#else /* __GNUC__ */ +static char *ident = NULL; + +void +nogcc (void) +{ + ident = NULL; +} + +#endif /* __GNUC__ */ diff --git a/sys/oss/gstosshelper.h b/sys/oss/gstosshelper.h new file mode 100644 index 00000000..819a21a1 --- /dev/null +++ b/sys/oss/gstosshelper.h @@ -0,0 +1,44 @@ +/* Evil evil evil hack to get OSS apps to cooperate with esd + * Copyright (C) 1998, 1999 Manish Singh <yosh@gimp.org> + * + * 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_OSSGST_HELPER_H__ +#define __GST_OSSGST_HELPER_H__ + +#define HELPER_MAGIC_IN 500 +#define HELPER_MAGIC_OUT 501 +#define HELPER_MAGIC_SNDFD 502 + +#define CMD_DATA 1 +#define CMD_FORMAT 2 + +typedef struct { + char id; + + union { + unsigned int length; + struct { + int format; + int stereo; + int rate; + } format; + } cmd; +} command; + + +#endif /* __GST_OSSGST_HELPER_H__ */ diff --git a/sys/oss/gstosssink.c b/sys/oss/gstosssink.c new file mode 100644 index 00000000..407efd1a --- /dev/null +++ b/sys/oss/gstosssink.c @@ -0,0 +1,626 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000 Wim Taymans <wim.taymans@chello.be> + * + * 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. + */ + + +#include <sys/types.h> +#include <sys/stat.h> +#include <sys/ioctl.h> +#include <fcntl.h> +#include <sys/soundcard.h> +#include <unistd.h> +#include <errno.h> + +#include <gstosssink.h> + +static GstElementDetails gst_osssink_details = { + "Audio Sink (OSS)", + "Sink/Audio", + "Output to a sound card via OSS", + VERSION, + "Erik Walthinsen <omega@cse.ogi.edu>, " + "Wim Taymans <wim.taymans@chello.be>", + "(C) 1999", +}; + +static void gst_osssink_class_init (GstOssSinkClass *klass); +static void gst_osssink_init (GstOssSink *osssink); +static void gst_osssink_finalize (GObject *object); + +static gboolean gst_osssink_open_audio (GstOssSink *sink); +static void gst_osssink_close_audio (GstOssSink *sink); +static void gst_osssink_sync_parms (GstOssSink *osssink); +static GstElementStateReturn gst_osssink_change_state (GstElement *element); +static GstPadNegotiateReturn gst_osssink_negotiate (GstPad *pad, GstCaps **caps, gpointer *user_data); + +static void gst_osssink_set_property (GObject *object, guint prop_id, const GValue *value, + GParamSpec *pspec); +static void gst_osssink_get_property (GObject *object, guint prop_id, GValue *value, + GParamSpec *pspec); + +static void gst_osssink_chain (GstPad *pad,GstBuffer *buf); + +/* OssSink signals and args */ +enum { + SIGNAL_HANDOFF, + LAST_SIGNAL +}; + +enum { + ARG_0, + ARG_DEVICE, + ARG_MUTE, + ARG_FORMAT, + ARG_CHANNELS, + ARG_FREQUENCY, + ARG_FRAGMENT, + ARG_BUFFER_SIZE + /* FILL ME */ +}; + +GST_PADTEMPLATE_FACTORY (osssink_sink_factory, + "sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "osssink_sink", + "audio/raw", + "format", GST_PROPS_STRING ("int"), // hack + "law", GST_PROPS_INT (0), + "endianness", GST_PROPS_INT (G_BYTE_ORDER), + "signed", GST_PROPS_LIST ( + GST_PROPS_BOOLEAN (FALSE), + GST_PROPS_BOOLEAN (TRUE) + ), + "width", GST_PROPS_LIST ( + GST_PROPS_INT (8), + GST_PROPS_INT (16) + ), + "depth", GST_PROPS_LIST ( + GST_PROPS_INT (8), + GST_PROPS_INT (16) + ), + "rate", GST_PROPS_INT_RANGE (8000, 48000), + "channels", GST_PROPS_INT_RANGE (1, 2) + ) +); + +#define GST_TYPE_OSSSINK_CHANNELS (gst_osssink_channels_get_type()) +static GType +gst_osssink_channels_get_type(void) { + static GType osssink_channels_type = 0; + static GEnumValue osssink_channels[] = { + {0, "0", "Silence"}, + {1, "1", "Mono"}, + {2, "2", "Stereo"}, + {0, NULL, NULL}, + }; + if (!osssink_channels_type) { + osssink_channels_type = g_enum_register_static("GstAudiosinkChannels", osssink_channels); + } + return osssink_channels_type; +} + + +static GstElementClass *parent_class = NULL; +static guint gst_osssink_signals[LAST_SIGNAL] = { 0 }; + +GType +gst_osssink_get_type (void) +{ + static GType osssink_type = 0; + + if (!osssink_type) { + static const GTypeInfo osssink_info = { + sizeof(GstOssSinkClass), + NULL, + NULL, + (GClassInitFunc)gst_osssink_class_init, + NULL, + NULL, + sizeof(GstOssSink), + 0, + (GInstanceInitFunc)gst_osssink_init, + }; + osssink_type = g_type_register_static (GST_TYPE_ELEMENT, "GstOssSink", &osssink_info, 0); + } + + return osssink_type; +} + +static GstBufferPool* +gst_osssink_get_bufferpool (GstPad *pad) +{ + GstOssSink *oss; + + oss = GST_OSSSINK (gst_pad_get_parent(pad)); + + return oss->sinkpool; +} + +static void +gst_osssink_finalize (GObject *object) +{ + GstOssSink *osssink = (GstOssSink *) object; + + g_free (osssink->device); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static void +gst_osssink_class_init (GstOssSinkClass *klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass*)klass; + gstelement_class = (GstElementClass*)klass; + + parent_class = g_type_class_ref(GST_TYPE_ELEMENT); + + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_DEVICE, + g_param_spec_string("device","device","device", + "/dev/dsp",G_PARAM_READWRITE)); // CHECKME! + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_MUTE, + g_param_spec_boolean("mute","mute","mute", + TRUE,G_PARAM_READWRITE)); + + // it would be nice to show format in symbolic form, oh well + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FORMAT, + g_param_spec_int ("format","format","format", + 0, G_MAXINT, AFMT_S16_LE, G_PARAM_READWRITE)); + + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_CHANNELS, + g_param_spec_enum("channels","channels","channels", + GST_TYPE_OSSSINK_CHANNELS,2,G_PARAM_READWRITE)); + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FREQUENCY, + g_param_spec_int("frequency","frequency","frequency", + 0,G_MAXINT,44100,G_PARAM_READWRITE)); + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FRAGMENT, + g_param_spec_int("fragment","fragment","fragment", + 0,G_MAXINT,6,G_PARAM_READWRITE)); + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BUFFER_SIZE, + g_param_spec_int("buffer_size","buffer_size","buffer_size", + 0,G_MAXINT,4096,G_PARAM_READWRITE)); + + gst_osssink_signals[SIGNAL_HANDOFF] = + g_signal_new("handoff",G_TYPE_FROM_CLASS(klass), G_SIGNAL_RUN_LAST, + G_STRUCT_OFFSET(GstOssSinkClass,handoff), NULL, NULL, + g_cclosure_marshal_VOID__VOID,G_TYPE_NONE,0); + + gobject_class->set_property = gst_osssink_set_property; + gobject_class->get_property = gst_osssink_get_property; + gobject_class->finalize = gst_osssink_finalize; + + gstelement_class->change_state = gst_osssink_change_state; +} + +static void +gst_osssink_init (GstOssSink *osssink) +{ + osssink->sinkpad = gst_pad_new_from_template ( + GST_PADTEMPLATE_GET (osssink_sink_factory), "sink"); + gst_element_add_pad (GST_ELEMENT (osssink), osssink->sinkpad); + gst_pad_set_negotiate_function (osssink->sinkpad, gst_osssink_negotiate); + gst_pad_set_bufferpool_function (osssink->sinkpad, gst_osssink_get_bufferpool); + + gst_pad_set_chain_function (osssink->sinkpad, gst_osssink_chain); + + osssink->device = g_strdup ("/dev/dsp"); + osssink->fd = -1; + osssink->clock = gst_clock_get_system(); + osssink->channels = 1; + osssink->frequency = 11025; + osssink->fragment = 6; +/* AFMT_*_BE not available on all OSS includes (e.g. FBSD) */ +#ifdef WORDS_BIGENDIAN + osssink->format = AFMT_S16_BE; +#else + osssink->format = AFMT_S16_LE; +#endif /* WORDS_BIGENDIAN */ + gst_clock_register (osssink->clock, GST_OBJECT (osssink)); + osssink->bufsize = 4096; + /* 6 buffers per chunk by default */ + osssink->sinkpool = gst_buffer_pool_get_default (osssink->bufsize, 6); + + GST_FLAG_SET (osssink, GST_ELEMENT_THREAD_SUGGESTED); +} + +static gboolean +gst_osssink_parse_caps (GstOssSink *osssink, GstCaps *caps) +{ + gint law, endianness, width, depth; + gboolean sign; + gint format = -1; + + // deal with the case where there are no props... + if (gst_caps_get_props(caps) == NULL) return FALSE; + + width = gst_caps_get_int (caps, "width"); + depth = gst_caps_get_int (caps, "depth"); + + if (width != depth) return FALSE; + + law = gst_caps_get_int (caps, "law"); + endianness = gst_caps_get_int (caps, "endianness"); + sign = gst_caps_get_boolean (caps, "signed"); + + if (law == 0) { + if (width == 16) { + if (sign == TRUE) { + if (endianness == G_LITTLE_ENDIAN) + format = AFMT_S16_LE; + else if (endianness == G_BIG_ENDIAN) + format = AFMT_S16_BE; + } + else { + if (endianness == G_LITTLE_ENDIAN) + format = AFMT_U16_LE; + else if (endianness == G_BIG_ENDIAN) + format = AFMT_U16_BE; + } + } + else if (width == 8) { + if (sign == TRUE) { + format = AFMT_S8; + } + else { + format = AFMT_U8; + } + } + } + + if (format == -1) + return FALSE; + + osssink->format = format; + osssink->channels = gst_caps_get_int (caps, "channels"); + osssink->frequency = gst_caps_get_int (caps, "rate"); + + return TRUE; +} + +static GstPadNegotiateReturn +gst_osssink_negotiate (GstPad *pad, GstCaps **caps, gpointer *user_data) +{ + GstOssSink *osssink; + + g_return_val_if_fail (pad != NULL, GST_PAD_NEGOTIATE_FAIL); + g_return_val_if_fail (GST_IS_PAD (pad), GST_PAD_NEGOTIATE_FAIL); + + osssink = GST_OSSSINK (gst_pad_get_parent (pad)); + + GST_INFO (GST_CAT_NEGOTIATION, "osssink: negotiate"); + // we decide + if (user_data == NULL) { + *caps = NULL; + return GST_PAD_NEGOTIATE_TRY; + } + // have we got caps? + else if (*caps) { + + if (gst_osssink_parse_caps (osssink, *caps)) { + gst_osssink_sync_parms (osssink); + + return GST_PAD_NEGOTIATE_AGREE; + } + + // FIXME check if the sound card was really set to these caps, + // else send out another caps.. + + return GST_PAD_NEGOTIATE_FAIL; + } + + return GST_PAD_NEGOTIATE_FAIL; +} + +static void +gst_osssink_sync_parms (GstOssSink *osssink) +{ + audio_buf_info ospace; + int frag; + + g_return_if_fail (osssink != NULL); + g_return_if_fail (GST_IS_OSSSINK (osssink)); + + if (osssink->fd == -1) return; + + if (osssink->fragment >> 16) + frag = osssink->fragment; + else + frag = 0x7FFF0000 | osssink->fragment; + + ioctl (osssink->fd, SNDCTL_DSP_SETFRAGMENT, &frag); + + ioctl (osssink->fd, SNDCTL_DSP_RESET, 0); + + ioctl (osssink->fd, SNDCTL_DSP_SETFMT, &osssink->format); + ioctl (osssink->fd, SNDCTL_DSP_CHANNELS, &osssink->channels); + ioctl (osssink->fd, SNDCTL_DSP_SPEED, &osssink->frequency); + + ioctl (osssink->fd, SNDCTL_DSP_GETBLKSIZE, &frag); + ioctl (osssink->fd, SNDCTL_DSP_GETOSPACE, &ospace); + + /* + g_warning ("osssink: setting sound card to %dHz %d bit %s (%d bytes buffer, %d fragment)\n", + osssink->frequency, osssink->format, + (osssink->channels == 2) ? "stereo" : "mono", ospace.bytes, frag); + */ + GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: setting sound card to %dHz %d bit %s (%d bytes buffer, %d fragment)", + osssink->frequency, osssink->format, + (osssink->channels == 2) ? "stereo" : "mono", ospace.bytes, frag); + +} + +static void +gst_osssink_chain (GstPad *pad, GstBuffer *buf) +{ + GstOssSink *osssink; + gboolean in_flush; + audio_buf_info ospace; + + g_return_if_fail (pad != NULL); + g_return_if_fail (GST_IS_PAD (pad)); + g_return_if_fail (buf != NULL); + + + /* this has to be an audio buffer */ + osssink = GST_OSSSINK (gst_pad_get_parent (pad)); +// g_return_if_fail(GST_FLAG_IS_SET(osssink,GST_STATE_RUNNING)); + + if (GST_IS_EVENT (buf)) { + g_print ("eos on osssink\n"); + gst_element_set_state (osssink, GST_STATE_PAUSED); + gst_event_free (GST_EVENT (buf)); + } + + if ((in_flush = GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLUSH))) { + GST_DEBUG (GST_CAT_PLUGIN_INFO,"osssink: flush\n"); + ioctl (osssink->fd, SNDCTL_DSP_RESET, 0); + } + + g_signal_emit (G_OBJECT (osssink), gst_osssink_signals[SIGNAL_HANDOFF], 0, + osssink); + + if (GST_BUFFER_DATA (buf) != NULL) { +#ifndef GST_DISABLE_TRACE + gst_trace_add_entry(NULL, 0, buf, "osssink: writing to soundcard"); +#endif // GST_DISABLE_TRACE + //g_print("osssink: writing to soundcard\n"); + if (osssink->fd >= 0) { + if (!osssink->mute) { + gst_clock_wait (osssink->clock, GST_BUFFER_TIMESTAMP (buf), GST_OBJECT (osssink)); + ioctl (osssink->fd, SNDCTL_DSP_GETOSPACE, &ospace); + GST_DEBUG (GST_CAT_PLUGIN_INFO,"osssink: (%d bytes buffer) %d %p %d\n", ospace.bytes, + osssink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); + write (osssink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); + //write(STDOUT_FILENO,GST_BUFFER_DATA(buf),GST_BUFFER_SIZE(buf)); + } + } + } + gst_buffer_unref (buf); +} + +static void +gst_osssink_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) +{ + GstOssSink *osssink; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_OSSSINK (object)); + + osssink = GST_OSSSINK (object); + + switch (prop_id) { + case ARG_DEVICE: + osssink->device = g_strdup (g_value_get_string (value)); + break; + case ARG_MUTE: + osssink->mute = g_value_get_boolean (value); + break; + case ARG_FORMAT: + osssink->format = g_value_get_int (value); + gst_osssink_sync_parms (osssink); + break; + case ARG_CHANNELS: + osssink->channels = g_value_get_enum (value); + gst_osssink_sync_parms (osssink); + break; + case ARG_FREQUENCY: + osssink->frequency = g_value_get_int (value); + gst_osssink_sync_parms (osssink); + break; + case ARG_FRAGMENT: + osssink->fragment = g_value_get_int (value); + gst_osssink_sync_parms (osssink); + break; + case ARG_BUFFER_SIZE: + osssink->bufsize = g_value_get_int (value); + osssink->sinkpool = gst_buffer_pool_get_default (osssink->bufsize, 6); + break; + default: + break; + } +} + +static void +gst_osssink_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) +{ + GstOssSink *osssink; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_OSSSINK (object)); + + osssink = GST_OSSSINK (object); + + switch (prop_id) { + case ARG_DEVICE: + g_value_set_string (value, osssink->device); + break; + case ARG_MUTE: + g_value_set_boolean (value, osssink->mute); + break; + case ARG_FORMAT: + g_value_set_int (value, osssink->format); + break; + case ARG_CHANNELS: + g_value_set_enum (value, osssink->channels); + break; + case ARG_FREQUENCY: + g_value_set_int (value, osssink->frequency); + break; + case ARG_FRAGMENT: + g_value_set_int (value, osssink->fragment); + break; + case ARG_BUFFER_SIZE: + g_value_set_int (value, osssink->bufsize); + break; + default: + break; + } +} + +static gboolean +gst_osssink_open_audio (GstOssSink *sink) +{ + gint caps; + g_return_val_if_fail (sink->fd == -1, FALSE); + + GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: attempting to open sound device"); + + /* first try to open the sound card */ + sink->fd = open(sink->device, O_WRONLY | O_NONBLOCK); + if (errno == EBUSY) { + g_warning ("osssink: unable to open the sound device (in use ?)\n"); + return FALSE; + } + + /* re-open the sound device in blocking mode */ + close(sink->fd); + sink->fd = open(sink->device, O_WRONLY); + + /* if we have it, set the default parameters and go have fun */ + if (sink->fd >= 0) { + /* set card state */ + ioctl(sink->fd, SNDCTL_DSP_GETCAPS, &caps); + + GST_INFO(GST_CAT_PLUGIN_INFO, "osssink: Capabilities %08x", caps); + + if (caps & DSP_CAP_DUPLEX) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Full duplex"); + if (caps & DSP_CAP_REALTIME) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Realtime"); + if (caps & DSP_CAP_BATCH) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Batch"); + if (caps & DSP_CAP_COPROC) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Has coprocessor"); + if (caps & DSP_CAP_TRIGGER) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Trigger"); + if (caps & DSP_CAP_MMAP) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Direct access"); + +#ifdef DSP_CAP_MULTI + if (caps & DSP_CAP_MULTI) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Multiple open"); +#endif /* DSP_CAP_MULTI */ + +#ifdef DSP_CAP_BIND + if (caps & DSP_CAP_BIND) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Channel binding"); +#endif /* DSP_CAP_BIND */ + + ioctl(sink->fd, SNDCTL_DSP_GETFMTS, &caps); + + GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: Formats %08x", caps); + if (caps & AFMT_MU_LAW) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: MU_LAW"); + if (caps & AFMT_A_LAW) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: A_LAW"); + if (caps & AFMT_IMA_ADPCM) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: IMA_ADPCM"); + if (caps & AFMT_U8) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: U8"); + if (caps & AFMT_S16_LE) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: S16_LE"); + if (caps & AFMT_S16_BE) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: S16_BE"); + if (caps & AFMT_S8) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: S8"); + if (caps & AFMT_U16_LE) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: U16_LE"); + if (caps & AFMT_U16_BE) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: U16_BE"); + if (caps & AFMT_MPEG) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: MPEG"); +#ifdef AFMT_AC3 + if (caps & AFMT_AC3) GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: AC3"); +#endif + + GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: opened audio (%s) with fd=%d", sink->device, sink->fd); + GST_FLAG_SET (sink, GST_OSSSINK_OPEN); + + gst_osssink_sync_parms (sink); + return TRUE; + } + + return FALSE; +} + +static void +gst_osssink_close_audio (GstOssSink *sink) +{ + if (sink->fd < 0) return; + + close(sink->fd); + sink->fd = -1; + + GST_FLAG_UNSET (sink, GST_OSSSINK_OPEN); + + GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: closed sound device"); +} + +static GstElementStateReturn +gst_osssink_change_state (GstElement *element) +{ + GstOssSink *osssink; + + g_return_val_if_fail (GST_IS_OSSSINK (element), FALSE); + + osssink = GST_OSSSINK (element); + + /* if going down into NULL state, close the file if it's open */ + if (GST_STATE_PENDING (element) == GST_STATE_NULL) { + if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN)) + gst_osssink_close_audio (osssink); + + /* otherwise (READY) we need to open the sound card */ + } else if (GST_STATE_PENDING (element) == GST_STATE_READY) { + if (!GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN)) { + if (!gst_osssink_open_audio (osssink)) { + return GST_STATE_FAILURE; + } + } + } + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; +} + +gboolean +gst_osssink_factory_init (GstPlugin *plugin) +{ + GstElementFactory *factory; + + factory = gst_elementfactory_new ("osssink", GST_TYPE_OSSSINK, &gst_osssink_details); + g_return_val_if_fail (factory != NULL, FALSE); + + gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (osssink_sink_factory)); + + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); + + return TRUE; +} + diff --git a/sys/oss/gstosssink.h b/sys/oss/gstosssink.h new file mode 100644 index 00000000..fa42324c --- /dev/null +++ b/sys/oss/gstosssink.h @@ -0,0 +1,96 @@ +/* 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 <config.h> +#include <gst/gst.h> + + +#ifdef __cplusplus +extern "C" { +#endif /* __cplusplus */ + + +#define GST_TYPE_OSSSINK \ + (gst_osssink_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(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSSINK)) + +typedef enum { + GST_OSSSINK_OPEN = GST_ELEMENT_FLAG_LAST, + + GST_OSSSINK_FLAG_LAST = GST_ELEMENT_FLAG_LAST+2, +} GstOssSinkFlags; + +typedef struct _GstOssSink GstOssSink; +typedef struct _GstOssSinkClass GstOssSinkClass; + +struct _GstOssSink { + GstElement element; + + GstPad *sinkpad; + GstBufferPool *sinkpool; + + //GstClockTime clocktime; + GstClock *clock; + + /* device */ + gchar *device; + + /* soundcard state */ + int fd; + int caps; /* the capabilities */ + gint format; + gint channels; + gint frequency; + gint fragment; + gboolean mute; + guint bufsize; +}; + +struct _GstOssSinkClass { + GstElementClass parent_class; + + /* signals */ + void (*handoff) (GstElement *element,GstPad *pad); +}; + +GType gst_osssink_get_type(void); + +gboolean gst_osssink_factory_init(GstPlugin *plugin); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + + +#endif /* __GST_OSSSINK_H__ */ diff --git a/sys/oss/gstosssrc.c b/sys/oss/gstosssrc.c new file mode 100644 index 00000000..ede0be37 --- /dev/null +++ b/sys/oss/gstosssrc.c @@ -0,0 +1,419 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2000 Wim Taymans <wtay@chello.be> + * + * 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. + */ + +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> +#include <sys/soundcard.h> +#include <sys/ioctl.h> +#include <unistd.h> + +#include <gstosssrc.h> + +//#define WEIRD_THING /* show the weird thing with 44100 KHz rec */ + +static GstElementDetails gst_osssrc_details = { + "Audio Source (OSS)", + "Source/Audio", + "Read from the sound card", + VERSION, + "Erik Walthinsen <omega@cse.ogi.edu>", + "(C) 1999", +}; + + +/* OssSrc signals and args */ +enum { + /* FILL ME */ + LAST_SIGNAL +}; + +enum { + ARG_0, + ARG_DEVICE, + ARG_BYTESPERREAD, + ARG_CUROFFSET, + ARG_FORMAT, + ARG_CHANNELS, + ARG_FREQUENCY +}; + +GST_PADTEMPLATE_FACTORY (osssrc_src_factory, + "src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "osssrc_src", + "audio/raw", + "format", GST_PROPS_STRING ("int"), + "law", GST_PROPS_INT (0), + "endianness", GST_PROPS_INT (G_BYTE_ORDER), + "signed", GST_PROPS_LIST ( + GST_PROPS_BOOLEAN (TRUE), + GST_PROPS_BOOLEAN (FALSE) + ), + "width", GST_PROPS_LIST ( + GST_PROPS_INT (8), + GST_PROPS_INT (16) + ), + "depth", GST_PROPS_LIST ( + GST_PROPS_INT (8), + GST_PROPS_INT (16) + ), + "rate", GST_PROPS_INT_RANGE (8000, 48000), + "channels", GST_PROPS_INT_RANGE (1, 2) + ) +) + +static void gst_osssrc_class_init (GstOssSrcClass *klass); +static void gst_osssrc_init (GstOssSrc *osssrc); + +static void gst_osssrc_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec); +static void gst_osssrc_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec); +static GstElementStateReturn gst_osssrc_change_state (GstElement *element); + +static void gst_osssrc_close_audio (GstOssSrc *src); +static gboolean gst_osssrc_open_audio (GstOssSrc *src); +static void gst_osssrc_sync_parms (GstOssSrc *osssrc); + +static GstBuffer * gst_osssrc_get (GstPad *pad); + +static GstElementClass *parent_class = NULL; +//static guint gst_osssrc_signals[LAST_SIGNAL] = { 0 }; + +GType +gst_osssrc_get_type (void) +{ + static GType osssrc_type = 0; + + if (!osssrc_type) { + static const GTypeInfo osssrc_info = { + sizeof(GstOssSrcClass), + NULL, + NULL, + (GClassInitFunc)gst_osssrc_class_init, + NULL, + NULL, + sizeof(GstOssSrc), + 0, + (GInstanceInitFunc)gst_osssrc_init, + }; + osssrc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstOssSrc", &osssrc_info, 0); + } + return osssrc_type; +} + +static void +gst_osssrc_class_init (GstOssSrcClass *klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass*)klass; + gstelement_class = (GstElementClass*)klass; + + parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BYTESPERREAD, + g_param_spec_ulong("bytes_per_read","bytes_per_read","bytes_per_read", + 0,G_MAXULONG,0,G_PARAM_READWRITE)); // CHECKME + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_CUROFFSET, + g_param_spec_ulong("curoffset","curoffset","curoffset", + 0,G_MAXULONG,0,G_PARAM_READABLE)); // CHECKME + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FORMAT, + g_param_spec_int("format","format","format", + G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); // CHECKME + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_CHANNELS, + g_param_spec_int("channels","channels","channels", + G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); // CHECKME + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FREQUENCY, + g_param_spec_int("frequency","frequency","frequency", + G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); // CHECKME + g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_DEVICE, + g_param_spec_string("device","device","oss device (/dev/dspN usually)", + "default",G_PARAM_READWRITE)); + + gobject_class->set_property = gst_osssrc_set_property; + gobject_class->get_property = gst_osssrc_get_property; + + gstelement_class->change_state = gst_osssrc_change_state; +} + +static void +gst_osssrc_init (GstOssSrc *osssrc) +{ + osssrc->srcpad = gst_pad_new_from_template ( + GST_PADTEMPLATE_GET (osssrc_src_factory), "src"); + gst_pad_set_get_function(osssrc->srcpad,gst_osssrc_get); + gst_element_add_pad (GST_ELEMENT (osssrc), osssrc->srcpad); + + osssrc->device = g_strdup ("/dev/dsp"); + osssrc->fd = -1; + + /* adding some default values */ + osssrc->format = AFMT_S16_LE; + osssrc->channels = 2; + osssrc->frequency = 44100; + + osssrc->bytes_per_read = 4096; + osssrc->curoffset = 0; + osssrc->seq = 0; +} + +static GstBuffer * +gst_osssrc_get (GstPad *pad) +{ + GstOssSrc *src; + GstBuffer *buf; + glong readbytes; + + g_return_val_if_fail (pad != NULL, NULL); + src = GST_OSSSRC(gst_pad_get_parent (pad)); + + GST_DEBUG (0, "attempting to read something from soundcard\n"); + + buf = gst_buffer_new (); + g_return_val_if_fail (buf, NULL); + + GST_BUFFER_DATA (buf) = (gpointer)g_malloc (src->bytes_per_read); + + readbytes = read (src->fd,GST_BUFFER_DATA (buf), + src->bytes_per_read); + + if (readbytes == 0) { + gst_element_signal_eos (GST_ELEMENT (src)); + return NULL; + } + + GST_BUFFER_SIZE (buf) = readbytes; + GST_BUFFER_OFFSET (buf) = src->curoffset; + + src->curoffset += readbytes; + + GST_DEBUG (0, "pushed buffer from soundcard of %ld bytes\n", readbytes); + return buf; +} + +static void +gst_osssrc_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) +{ + GstOssSrc *src; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_OSSSRC (object)); + + src = GST_OSSSRC (object); + + switch (prop_id) { + case ARG_BYTESPERREAD: + src->bytes_per_read = g_value_get_ulong (value); + break; + case ARG_FORMAT: + src->format = g_value_get_int (value); + break; + case ARG_CHANNELS: + src->channels = g_value_get_int (value); + break; + case ARG_FREQUENCY: + src->frequency = g_value_get_int (value); + break; + case ARG_CUROFFSET: + src->curoffset = g_value_get_int (value); + break; + case ARG_DEVICE: + g_free(src->device); + src->device = g_strdup (g_value_get_string (value)); + break; + default: + break; + } +} + +static void +gst_osssrc_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec) +{ + GstOssSrc *src; + + /* it's not null if we got it, but it might not be ours */ + g_return_if_fail (GST_IS_OSSSRC (object)); + + src = GST_OSSSRC (object); + + switch (prop_id) { + case ARG_BYTESPERREAD: + g_value_set_ulong (value, src->bytes_per_read); + break; + case ARG_FORMAT: + g_value_set_int (value, src->format); + break; + case ARG_CHANNELS: + g_value_set_int (value, src->channels); + break; + case ARG_FREQUENCY: + g_value_set_int (value, src->frequency); + break; + case ARG_CUROFFSET: + g_value_set_ulong (value, src->curoffset); + break; + case ARG_DEVICE: + g_value_set_string (value, src->device); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstElementStateReturn +gst_osssrc_change_state (GstElement *element) +{ + g_return_val_if_fail (GST_IS_OSSSRC (element), FALSE); + GST_DEBUG (0, "osssrc: state change\n"); + /* if going down into NULL state, close the file if it's open */ + if (GST_STATE_PENDING (element) == GST_STATE_NULL) { + if (GST_FLAG_IS_SET (element, GST_OSSSRC_OPEN)) + gst_osssrc_close_audio (GST_OSSSRC (element)); + /* otherwise (READY or higher) we need to open the sound card */ + } else { + GST_DEBUG (0, "DEBUG: osssrc: ready or higher\n"); + + if (!GST_FLAG_IS_SET (element, GST_OSSSRC_OPEN)) { + if (!gst_osssrc_open_audio (GST_OSSSRC (element))) + return GST_STATE_FAILURE; + } + } + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; +} + +static gboolean +gst_osssrc_open_audio (GstOssSrc *src) +{ + g_return_val_if_fail (!GST_FLAG_IS_SET (src, GST_OSSSRC_OPEN), FALSE); + + /* first try to open the sound card */ + src->fd = open(src->device, O_RDONLY); + + /* if we have it, set the default parameters and go have fun */ + if (src->fd > 0) { + + /* set card state */ + gst_osssrc_sync_parms (src); + GST_DEBUG (0,"opened audio: %s\n",src->device); + + GST_FLAG_SET (src, GST_OSSSRC_OPEN); + return TRUE; + } + + return FALSE; +} + +static void +gst_osssrc_close_audio (GstOssSrc *src) +{ + g_return_if_fail (GST_FLAG_IS_SET (src, GST_OSSSRC_OPEN)); + + close(src->fd); + src->fd = -1; + + GST_FLAG_UNSET (src, GST_OSSSRC_OPEN); +} + +static void +gst_osssrc_sync_parms (GstOssSrc *osssrc) +{ + audio_buf_info ispace; + gint frag; + /* hack : ioctl on frequency seems to *change* the frequency + maybe this is right, but at least for 44100 Khz, it + gives unexpected results ! */ + + guint frequency; + + g_return_if_fail (osssrc != NULL); + g_return_if_fail (GST_IS_OSSSRC (osssrc)); + g_return_if_fail (osssrc->fd > 0); + + frequency = osssrc->frequency; + + frag = 0x7fff0006; + + ioctl(osssrc->fd, SNDCTL_DSP_SETFRAGMENT, &frag); + ioctl(osssrc->fd, SNDCTL_DSP_RESET, 0); + + ioctl(osssrc->fd, SNDCTL_DSP_SETFMT, &osssrc->format); + ioctl(osssrc->fd, SNDCTL_DSP_CHANNELS, &osssrc->channels); +#ifdef WEIRD_THING + g_print ("DEBUG: do you want to see A Weird Thing (TM) ?\n"); + g_print ("DEBUG: then set osssrc to a 44100 frequency and check these numbers.\n"); + g_print ("DEBUG: osssrc frequency before ioctl: %d\n", osssrc->frequency); +#endif + ioctl(osssrc->fd, SNDCTL_DSP_SPEED, &osssrc->frequency); +#ifdef WEIRD_THING + g_print ("DEBUG: osssrc frequency after ioctl : %d\n", osssrc->frequency); + g_print ("DEBUG: redoing ioctl using a temp variable, and resetting osssrc->frequency\n"); + osssrc->frequency = frequency; + ioctl(osssrc->fd, SNDCTL_DSP_SPEED, &frequency); +#endif + ioctl(osssrc->fd, SNDCTL_DSP_GETISPACE, &ispace); + ioctl(osssrc->fd, SNDCTL_DSP_GETBLKSIZE, &frag); + + g_print("setting sound card to %dKHz %d bit %s (%d bytes buffer, %d fragment)\n", + osssrc->frequency, osssrc->format, + (osssrc->channels == 2) ? "stereo" : "mono", ispace.bytes, frag); + + /* set caps on src pad */ + gst_pad_set_caps (osssrc->srcpad, gst_caps_new ( + "oss_src", + "audio/raw", + gst_props_new ( + "format", GST_PROPS_STRING ("int"), + "law", GST_PROPS_INT (0), //FIXME + "endianness", GST_PROPS_INT (G_BYTE_ORDER), //FIXME + "signed", GST_PROPS_BOOLEAN (TRUE), //FIXME + "width", GST_PROPS_INT (osssrc->format), + "depth", GST_PROPS_INT (osssrc->format), + "rate", GST_PROPS_INT (osssrc->frequency), + "channels", GST_PROPS_INT (osssrc->channels), + NULL + ) + )); +} + +gboolean +gst_osssrc_factory_init (GstPlugin *plugin) +{ + GstElementFactory *factory; + + factory = gst_elementfactory_new ("osssrc", GST_TYPE_OSSSRC, &gst_osssrc_details); + g_return_val_if_fail (factory != NULL, FALSE); + + gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (osssrc_src_factory)); + + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); + + return TRUE; +} + diff --git a/sys/oss/gstosssrc.h b/sys/oss/gstosssrc.h new file mode 100644 index 00000000..21181100 --- /dev/null +++ b/sys/oss/gstosssrc.h @@ -0,0 +1,94 @@ +/* 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_OSSSRC_H__ +#define __GST_OSSSRC_H__ + + +#include <config.h> +#include <gst/gst.h> + + +#ifdef __cplusplus +extern "C" { +#endif /* __cplusplus */ + + +#define GST_TYPE_OSSSRC \ + (gst_osssrc_get_type()) +#define GST_OSSSRC(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSSRC,GstOssSrc)) +#define GST_OSSSRC_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSSRC,GstOssSrcClass)) +#define GST_IS_OSSSRC(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSSRC)) +#define GST_IS_OSSSRC_CLASS(obj) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSSRC)) + +typedef enum { + GST_OSSSRC_OPEN = GST_ELEMENT_FLAG_LAST, + + GST_OSSSRC_FLAG_LAST = GST_ELEMENT_FLAG_LAST+2, +} GstOssSrcFlags; + +typedef struct _GstOssSrc GstOssSrc; +typedef struct _GstOssSrcClass GstOssSrcClass; + +struct _GstOssSrc { + GstElement element; + + /* pads */ + GstPad *srcpad; + + /* device */ + gchar *device; + + /* sound card */ + gint fd; + + /* audio parameters */ + gint format; + gint channels; + gint frequency; + + /* blocking */ + gulong curoffset; + gulong bytes_per_read; + + gulong seq; +}; + +struct _GstOssSrcClass { + GstElementClass parent_class; +}; + +GType gst_osssrc_get_type(void); + +gboolean gst_osssrc_factory_init (GstPlugin *plugin); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + + +#endif /* __GST_OSSSRC_H__ */ |