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gstreamer/sys/oss4/oss4-audio.c

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/* 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);
}
}
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);
}
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)
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