/* GStreamer OSS4 audio plugin * Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net> * * 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. */ /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray * with newer GLib versions (>= 2.31.0) */ #define GLIB_DISABLE_DEPRECATION_WARNINGS #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <sys/ioctl.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <unistd.h> #include <string.h> #include "gst/gst-i18n-plugin.h" #include <gst/audio/multichannel.h> #include "oss4-audio.h" #include "oss4-mixer.h" #include "oss4-property-probe.h" #include "oss4-sink.h" #include "oss4-source.h" #include "oss4-soundcard.h" GST_DEBUG_CATEGORY (oss4mixer_debug); GST_DEBUG_CATEGORY (oss4sink_debug); GST_DEBUG_CATEGORY (oss4src_debug); GST_DEBUG_CATEGORY (oss4_debug); #define GST_CAT_DEFAULT oss4_debug typedef struct { const GstBufferFormat gst_fmt; const gint oss_fmt; const gchar name[16]; const gint depth; const gint width; const gint endianness; const gboolean signedness; } GstOss4AudioFormat; /* *INDENT-OFF* */ static const GstOss4AudioFormat fmt_map[] = { /* note: keep sorted by preference, prefered formats first */ { GST_MU_LAW, AFMT_MU_LAW, "audio/x-mulaw", 0, 0, 0, FALSE}, { GST_A_LAW, AFMT_A_LAW, "audio/x-alaw", 0, 0, 0, FALSE}, { GST_S32_LE, AFMT_S32_LE, "audio/x-raw-int", 32, 32, G_LITTLE_ENDIAN, TRUE}, { GST_S32_BE, AFMT_S32_BE, "audio/x-raw-int", 32, 32, G_BIG_ENDIAN, TRUE}, { GST_S24_LE, AFMT_S24_LE, "audio/x-raw-int", 24, 32, G_LITTLE_ENDIAN, TRUE}, { GST_S24_BE, AFMT_S24_BE, "audio/x-raw-int", 24, 32, G_BIG_ENDIAN, TRUE}, { GST_S24_3LE, AFMT_S24_PACKED, "audio/x-raw-int", 24, 24, G_LITTLE_ENDIAN, TRUE}, { GST_S16_LE, AFMT_S16_LE, "audio/x-raw-int", 16, 16, G_LITTLE_ENDIAN, TRUE}, { GST_S16_BE, AFMT_S16_BE, "audio/x-raw-int", 16, 16, G_BIG_ENDIAN, TRUE}, { GST_U16_LE, AFMT_U16_LE, "audio/x-raw-int", 16, 16, G_LITTLE_ENDIAN, FALSE}, { GST_U16_BE, AFMT_U16_BE, "audio/x-raw-int", 16, 16, G_BIG_ENDIAN, FALSE}, { GST_S8, AFMT_S8, "audio/x-raw-int", 8, 8, 0, TRUE}, { GST_U8, AFMT_U8, "audio/x-raw-int", 8, 8, 0, FALSE} }; /* *INDENT-ON* */ /* formats we assume the OSS4 layer can always handle and convert internally */ #define CONVERTIBLE_FORMATS ( \ AFMT_MU_LAW | AFMT_A_LAW | \ AFMT_S32_LE | AFMT_S32_BE | \ AFMT_S24_LE | AFMT_S24_BE | \ AFMT_S24_PACKED | \ AFMT_S16_LE | AFMT_S16_BE | \ AFMT_U16_LE | AFMT_U16_BE | \ AFMT_S8 | AFMT_U8 ) static void gst_oss4_append_format_to_caps (const GstOss4AudioFormat * fmt, GstCaps * caps) { GstStructure *s; s = gst_structure_empty_new (fmt->name); if (fmt->width != 0 && fmt->depth != 0) { gst_structure_set (s, "width", G_TYPE_INT, fmt->width, "depth", G_TYPE_INT, fmt->depth, "signed", G_TYPE_BOOLEAN, fmt->signedness, NULL); } if (fmt->endianness != 0) { gst_structure_set (s, "endianness", G_TYPE_INT, fmt->endianness, NULL); } gst_caps_append_structure (caps, s); } static gint gst_oss4_audio_get_oss_format (GstBufferFormat fmt) { guint i; for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) { if (fmt_map[i].gst_fmt == fmt) return fmt_map[i].oss_fmt; } return 0; } /* These are pretty random */ #define GST_OSS4_MIN_SAMPLE_RATE 1 #define GST_OSS4_MAX_SAMPLE_RATE 192000 static gboolean gst_oss4_audio_detect_rates (GstObject * obj, oss_audioinfo * ai, GstCaps * caps) { GValue val = { 0, }; int minrate, maxrate, i; minrate = ai->min_rate; maxrate = ai->max_rate; /* sanity check */ if (minrate > maxrate) { GST_WARNING_OBJECT (obj, "min_rate %d > max_rate %d (buggy driver?)", minrate, maxrate); maxrate = ai->min_rate; /* swap */ minrate = ai->max_rate; } /* limit to something sensible */ if (minrate < GST_OSS4_MIN_SAMPLE_RATE) minrate = GST_OSS4_MIN_SAMPLE_RATE; if (maxrate > GST_OSS4_MAX_SAMPLE_RATE) maxrate = GST_OSS4_MAX_SAMPLE_RATE; if (maxrate < GST_OSS4_MIN_SAMPLE_RATE) { GST_WARNING_OBJECT (obj, "max_rate < %d, which makes no sense", GST_OSS4_MIN_SAMPLE_RATE); return FALSE; } GST_LOG_OBJECT (obj, "min_rate %d, max_rate %d (originally: %d, %d)", minrate, maxrate, ai->min_rate, ai->max_rate); if ((ai->caps & PCM_CAP_FREERATE)) { GST_LOG_OBJECT (obj, "device supports any sample rate between min and max"); if (minrate == maxrate) { g_value_init (&val, G_TYPE_INT); g_value_set_int (&val, maxrate); } else { g_value_init (&val, GST_TYPE_INT_RANGE); gst_value_set_int_range (&val, minrate, maxrate); } } else { GST_LOG_OBJECT (obj, "%d sample rates:", ai->nrates); g_value_init (&val, GST_TYPE_LIST); for (i = 0; i < ai->nrates; ++i) { GST_LOG_OBJECT (obj, " rate: %d", ai->rates[i]); if (ai->rates[i] >= minrate && ai->rates[i] <= maxrate) { GValue rate_val = { 0, }; g_value_init (&rate_val, G_TYPE_INT); g_value_set_int (&rate_val, ai->rates[i]); gst_value_list_append_value (&val, &rate_val); g_value_unset (&rate_val); } } if (gst_value_list_get_size (&val) == 0) { g_value_unset (&val); return FALSE; } } for (i = 0; i < gst_caps_get_size (caps); ++i) { GstStructure *s; s = gst_caps_get_structure (caps, i); gst_structure_set_value (s, "rate", &val); } g_value_unset (&val); return TRUE; } static void gst_oss4_audio_add_channel_layout (GstObject * obj, guint64 layout, guint num_channels, GstStructure * s) { const GstAudioChannelPosition pos_map[16] = { GST_AUDIO_CHANNEL_POSITION_NONE, /* 0 = dunno */ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, /* 1 = left */ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, /* 2 = right */ GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, /* 3 = center */ GST_AUDIO_CHANNEL_POSITION_LFE, /* 4 = lfe */ GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, /* 5 = left surround */ GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT, /* 6 = right surround */ GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, /* 7 = left rear */ GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, /* 8 = right rear */ GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE }; GstAudioChannelPosition ch_layout[8] = { 0, }; guint speaker_pos; /* speaker position as defined by OSS */ guint i; g_return_if_fail (num_channels <= G_N_ELEMENTS (ch_layout)); for (i = 0; i < num_channels; ++i) { /* layout contains up to 16 speaker positions, with each taking up 4 bits */ speaker_pos = (guint) ((layout >> (i * 4)) & 0x0f); /* if it's a channel position that's unknown to us, set all to NONE and * bail out */ if (G_UNLIKELY (pos_map[speaker_pos] == GST_AUDIO_CHANNEL_POSITION_NONE)) goto no_layout; ch_layout[i] = pos_map[speaker_pos]; } gst_audio_set_channel_positions (s, ch_layout); return; no_layout: { /* only warn if it's really unknown, position 0 is ok and represents NONE * (in which case we also just set all others to NONE ignoring the other * positions in the OSS-given layout, because that's what we currently * require in GStreamer) */ if (speaker_pos != 0) { GST_WARNING_OBJECT (obj, "unknown OSS channel position %x", ch_layout[i]); } for (i = 0; i < num_channels; ++i) { ch_layout[i] = GST_AUDIO_CHANNEL_POSITION_NONE; } gst_audio_set_channel_positions (s, ch_layout); return; } } /* arbitrary max. limit */ #define GST_OSS4_MIN_CHANNELS 1 #define GST_OSS4_MAX_CHANNELS 4096 /* takes ownership of the input caps */ static GstCaps * gst_oss4_audio_detect_channels (GstObject * obj, int fd, oss_audioinfo * ai, GstCaps * in_caps) { const gchar *forced_layout; GstStructure *s = NULL; guint64 layout = 0; GstCaps *chan_caps = NULL; GstCaps *out_caps = NULL; int minchans, maxchans; int c, i, j; /* GST_OSS4_CHANNEL_LAYOUT environment variable: may be used to force a * particular channel layout (if it contains an odd number of channel * positions it will also make us advertise a channel layout for that * channel count, even if we'd usually skip it; this is especially useful * for folks with 2.1 speakers, I guess) */ forced_layout = g_getenv ("GST_OSS4_CHANNEL_LAYOUT"); minchans = ai->min_channels; maxchans = ai->max_channels; /* sanity check */ if (minchans > maxchans) { GST_WARNING_OBJECT (obj, "min_chans %d > max_chans %d (buggy driver?)", minchans, maxchans); maxchans = ai->min_channels; /* swap */ minchans = ai->max_channels; } /* limit to something sensible */ if (minchans < GST_OSS4_MIN_CHANNELS) minchans = GST_OSS4_MIN_CHANNELS; if (maxchans > GST_OSS4_MAX_CHANNELS) maxchans = GST_OSS4_MAX_CHANNELS; if (maxchans < GST_OSS4_MIN_CHANNELS) { GST_WARNING_OBJECT (obj, "max_chans < %d, which makes no sense", GST_OSS4_MIN_CHANNELS); gst_caps_unref (in_caps); return NULL; } GST_LOG_OBJECT (obj, "min_channels %d, max_channels %d (originally: %d, %d)", minchans, maxchans, ai->min_channels, ai->max_channels); chan_caps = gst_caps_new_empty (); /* first do the simple cases: mono + stereo (channel layout implied) */ if (minchans == 1 && maxchans == 1) s = gst_structure_new ("x", "channels", G_TYPE_INT, 1, NULL); else if (minchans == 2 && maxchans >= 2) s = gst_structure_new ("x", "channels", G_TYPE_INT, 2, NULL); else if (minchans == 1 && maxchans >= 2) s = gst_structure_new ("x", "channels", GST_TYPE_INT_RANGE, 1, 2, NULL); gst_caps_append_structure (chan_caps, s); s = NULL; /* TODO: we assume all drivers use a left/right layout for stereo here */ if (maxchans <= 2) goto done; if (ioctl (fd, SNDCTL_DSP_GET_CHNORDER, &layout) == -1) { GST_WARNING_OBJECT (obj, "couldn't query channel layout, assuming default"); layout = CHNORDER_NORMAL; } GST_DEBUG_OBJECT (obj, "channel layout: %08" G_GINT64_MODIFIER "x", layout); /* e.g. forced 2.1 layout would be GST_OSS4_CHANNEL_LAYOUT=421 */ if (forced_layout != NULL && *forced_layout != '\0') { guint layout_len; layout_len = strlen (forced_layout); if (layout_len >= minchans && layout_len <= maxchans) { layout = g_ascii_strtoull (forced_layout, NULL, 16); maxchans = layout_len; GST_DEBUG_OBJECT (obj, "forced channel layout: %08" G_GINT64_MODIFIER "x" " ('%s'), maxchans now %d", layout, forced_layout, maxchans); } else { GST_WARNING_OBJECT (obj, "ignoring forced channel layout: layout has %d " "channel positions but maxchans is %d", layout_len, maxchans); } } /* need to advertise channel layouts for anything >2 and <=8 channels */ for (c = MAX (3, minchans); c <= MIN (maxchans, 8); c++) { /* "The min_channels and max_channels fields define the limits for the * number of channels. However some devices don't support all channels * within this range. It's possible that the odd values (3, 5, 7, 9, etc). * are not supported. There is currently no way to check for this other * than checking if SNDCTL_DSP_CHANNELS accepts the requested value. * Another approach is trying to avoid using odd number of channels." * * So, we don't know for sure if these odd values are supported: */ if ((c == 3 || c == 5 || c == 7) && (c != maxchans)) { GST_LOG_OBJECT (obj, "not adding layout with %d channels", c); continue; } s = gst_structure_new ("x", "channels", G_TYPE_INT, c, NULL); gst_oss4_audio_add_channel_layout (obj, layout, c, s); GST_LOG_OBJECT (obj, "c=%u, appending struct %" GST_PTR_FORMAT, c, s); gst_caps_append_structure (chan_caps, s); s = NULL; } if (maxchans <= 8) goto done; /* for everything >8 channels, CHANNEL_POSITION_NONE is implied. */ if (minchans == maxchans || maxchans == 9) { s = gst_structure_new ("x", "channels", G_TYPE_INT, maxchans, NULL); } else { s = gst_structure_new ("x", "channels", GST_TYPE_INT_RANGE, MAX (9, minchans), maxchans, NULL); } gst_caps_append_structure (chan_caps, s); s = NULL; done: GST_LOG_OBJECT (obj, "channel structures: %" GST_PTR_FORMAT, chan_caps); out_caps = gst_caps_new_empty (); /* combine each structure in the input caps with each channel caps struct */ for (i = 0; i < gst_caps_get_size (in_caps); ++i) { const GstStructure *in_s; in_s = gst_caps_get_structure (in_caps, i); for (j = 0; j < gst_caps_get_size (chan_caps); ++j) { const GstStructure *chan_s; const GValue *val; s = gst_structure_copy (in_s); chan_s = gst_caps_get_structure (chan_caps, j); if ((val = gst_structure_get_value (chan_s, "channels"))) gst_structure_set_value (s, "channels", val); if ((val = gst_structure_get_value (chan_s, "channel-positions"))) gst_structure_set_value (s, "channel-positions", val); gst_caps_append_structure (out_caps, s); s = NULL; } } gst_caps_unref (in_caps); gst_caps_unref (chan_caps); return out_caps; } GstCaps * gst_oss4_audio_probe_caps (GstObject * obj, int fd) { oss_audioinfo ai = { 0, }; gboolean output; GstCaps *caps; int nonnative_formats = 0; int formats, i; output = GST_IS_OSS4_SINK (obj); /* -1 = get info for currently open device (fd). This will fail with * OSS build <= 1013 because of a bug in OSS */ ai.dev = -1; if (ioctl (fd, SNDCTL_ENGINEINFO, &ai) == -1) goto engineinfo_failed; formats = (output) ? ai.oformats : ai.iformats; GST_LOG_OBJECT (obj, "%s formats : 0x%08x", (output) ? "out" : "in", formats); caps = gst_caps_new_empty (); /* first list all the formats natively supported */ for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) { if ((formats & fmt_map[i].oss_fmt)) { gst_oss4_append_format_to_caps (&fmt_map[i], caps); } else if ((fmt_map[i].oss_fmt & CONVERTIBLE_FORMATS)) { nonnative_formats |= fmt_map[i].oss_fmt; } } GST_LOG_OBJECT (obj, "adding non-native %s formats : 0x%08x", (output) ? "out" : "in", nonnative_formats); /* now append non-native formats for which conversion would be needed */ for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) { if ((nonnative_formats & fmt_map[i].oss_fmt)) { gst_oss4_append_format_to_caps (&fmt_map[i], caps); } } caps = gst_caps_do_simplify (caps); GST_LOG_OBJECT (obj, "formats: %" GST_PTR_FORMAT, caps); if (!gst_oss4_audio_detect_rates (obj, &ai, caps)) goto detect_rates_failed; caps = gst_oss4_audio_detect_channels (obj, fd, &ai, caps); if (caps == NULL) goto detect_channels_failed; GST_LOG_OBJECT (obj, "probed caps: %" GST_PTR_FORMAT, caps); return caps; /* ERRORS */ engineinfo_failed: { GST_WARNING ("ENGINEINFO supported formats probe failed: %s", g_strerror (errno)); return NULL; } detect_rates_failed: { GST_WARNING_OBJECT (obj, "failed to detect supported sample rates"); gst_caps_unref (caps); return NULL; } detect_channels_failed: { GST_WARNING_OBJECT (obj, "failed to detect supported channels"); gst_caps_unref (caps); return NULL; } } GstCaps * gst_oss4_audio_get_template_caps (void) { GstCaps *caps; gint i; caps = gst_caps_new_empty (); for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) { gst_oss4_append_format_to_caps (&fmt_map[i], caps); } caps = gst_caps_do_simplify (caps); for (i = 0; i < gst_caps_get_size (caps); ++i) { GstStructure *s; s = gst_caps_get_structure (caps, i); gst_structure_set (s, "rate", GST_TYPE_INT_RANGE, GST_OSS4_MIN_SAMPLE_RATE, GST_OSS4_MAX_SAMPLE_RATE, "channels", GST_TYPE_INT_RANGE, GST_OSS4_MIN_CHANNELS, GST_OSS4_MAX_CHANNELS, NULL); } return caps; } /* called by gst_oss4_sink_prepare() and gst_oss4_source_prepare() */ gboolean gst_oss4_audio_set_format (GstObject * obj, int fd, GstRingBufferSpec * spec) { struct audio_buf_info info = { 0, }; int fmt, chans, rate; fmt = gst_oss4_audio_get_oss_format (spec->format); if (fmt == 0) goto wrong_format; if (spec->type == GST_BUFTYPE_LINEAR && spec->width != 32 && spec->width != 24 && spec->width != 16 && spec->width != 8) { goto dodgy_width; } /* format */ GST_LOG_OBJECT (obj, "setting format: %d", fmt); if (ioctl (fd, SNDCTL_DSP_SETFMT, &fmt) == -1) goto set_format_failed; /* channels */ GST_LOG_OBJECT (obj, "setting channels: %d", spec->channels); chans = spec->channels; if (ioctl (fd, SNDCTL_DSP_CHANNELS, &chans) == -1) goto set_channels_failed; /* rate */ GST_LOG_OBJECT (obj, "setting rate: %d", spec->rate); rate = spec->rate; if (ioctl (fd, SNDCTL_DSP_SPEED, &rate) == -1) goto set_rate_failed; GST_DEBUG_OBJECT (obj, "effective format : %d", fmt); GST_DEBUG_OBJECT (obj, "effective channels : %d", chans); GST_DEBUG_OBJECT (obj, "effective rate : %d", rate); /* make sure format, channels, and rate are the ones we requested */ if (fmt != gst_oss4_audio_get_oss_format (spec->format) || chans != spec->channels || rate != spec->rate) { /* This shouldn't happen, but hey */ goto format_not_what_was_requested; } if (GST_IS_OSS4_SOURCE (obj)) { if (ioctl (fd, SNDCTL_DSP_GETISPACE, &info) == -1) goto get_ispace_failed; } else { if (ioctl (fd, SNDCTL_DSP_GETOSPACE, &info) == -1) goto get_ospace_failed; } spec->segsize = info.fragsize; /* we add some extra fragments -- this helps us account for delays due to * conversion buffer, streams queueing, etc. It is important that these * be taken into account because otherwise the delay counter can wind up * being too large, and the buffer will wrap. */ spec->segtotal = info.fragstotal + 4; spec->bytes_per_sample = (spec->width / 8) * spec->channels; GST_DEBUG_OBJECT (obj, "got segsize: %d, segtotal: %d, value: %08x", spec->segsize, spec->segtotal, info.fragsize); return TRUE; /* ERRORS */ wrong_format: { GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL), ("Unable to get format %d", spec->format)); return FALSE; } dodgy_width: { GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL), ("unexpected width %d", spec->width)); return FALSE; } set_format_failed: { GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL), ("DSP_SETFMT(%d) failed: %s", fmt, g_strerror (errno))); return FALSE; } set_channels_failed: { GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL), ("DSP_CHANNELS(%d) failed: %s", chans, g_strerror (errno))); return FALSE; } set_rate_failed: { GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL), ("DSP_SPEED(%d) failed: %s", rate, g_strerror (errno))); return FALSE; } get_ospace_failed: { GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL), ("DSP_GETOSPACE failed: %s", g_strerror (errno))); return FALSE; } get_ispace_failed: { GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL), ("DSP_GETISPACE failed: %s", g_strerror (errno))); return FALSE; } format_not_what_was_requested: { GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL), ("Format actually configured wasn't the one we requested. This is " "probably either a bug in the driver or in the format probing code.")); return FALSE; } } int gst_oss4_audio_get_version (GstObject * obj, int fd) { gint ver = 0; /* we use the old ioctl here on purpose instead of SNDCTL_SYSINFO */ if (ioctl (fd, OSS_GETVERSION, &ver) < 0) { GST_LOG_OBJECT (obj, "OSS_GETVERSION failed: %s", g_strerror (errno)); return -1; } GST_LOG_OBJECT (obj, "OSS version: 0x%08x", ver); return ver; } gboolean gst_oss4_audio_check_version (GstObject * obj, int fd) { return (gst_oss4_audio_get_version (obj, fd) >= GST_MIN_OSS4_VERSION); } gchar * gst_oss4_audio_find_device (GstObject * oss) { GValueArray *arr; gchar *ret = NULL; arr = gst_property_probe_probe_and_get_values_name (GST_PROPERTY_PROBE (oss), "device"); if (arr != NULL) { if (arr->n_values > 0) { const GValue *val; val = g_value_array_get_nth (arr, 0); ret = g_value_dup_string (val); } g_value_array_free (arr); } GST_LOG_OBJECT (oss, "first device found: %s", GST_STR_NULL (ret)); return ret; } static gboolean plugin_init (GstPlugin * plugin) { gint rank; GST_DEBUG_CATEGORY_INIT (oss4sink_debug, "oss4sink", 0, "OSS4 audio sink"); GST_DEBUG_CATEGORY_INIT (oss4src_debug, "oss4src", 0, "OSS4 audio src"); GST_DEBUG_CATEGORY_INIT (oss4mixer_debug, "oss4mixer", 0, "OSS4 mixer"); GST_DEBUG_CATEGORY_INIT (oss4_debug, "oss4", 0, "OSS4 plugin"); #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 /* we want a higher rank than the legacy OSS elements have now */ rank = GST_RANK_SECONDARY + 1; if (!gst_element_register (plugin, "oss4sink", rank, GST_TYPE_OSS4_SINK) || !gst_element_register (plugin, "oss4src", rank, GST_TYPE_OSS4_SOURCE) || !gst_element_register (plugin, "oss4mixer", rank, GST_TYPE_OSS4_MIXER)) { return FALSE; } return TRUE; } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, oss4, "Open Sound System (OSS) version 4 support for GStreamer", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)