--- /dev/null
+plugin_LTLIBRARIES = libgstoss4audio.la
+
+libgstoss4audio_la_SOURCES = \
+ oss4-audio.c \
+ oss4-mixer.c \
+ oss4-mixer-enum.c \
+ oss4-mixer-slider.c \
+ oss4-mixer-switch.c \
+ oss4-property-probe.c \
+ oss4-sink.c \
+ oss4-source.c
+
+libgstoss4audio_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
+libgstoss4audio_la_LIBADD = \
+ $(GST_PLUGINS_BASE_LIBS) \
+ -lgstinterfaces-$(GST_MAJORMINOR) \
+ -lgstaudio-$(GST_MAJORMINOR) \
+ $(GST_LIBS)
+libgstoss4audio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = \
+ oss4-audio.h \
+ oss4-mixer.h \
+ oss4-mixer-enum.h \
+ oss4-mixer-slider.h \
+ oss4-mixer-switch.h \
+ oss4-property-probe.h \
+ oss4-sink.h \
+ oss4-soundcard.h \
+ oss4-source.h
+
+
--- /dev/null
+/* 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.
+ */
+
+#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
+
+static const 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;
+} 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}
+};
+
+static gboolean
+gst_oss4_append_format_to_caps (gint fmt, GstCaps * caps)
+{
+ gint i;
+
+ for (i = 0; i < G_N_ELEMENTS (fmt_map); ++i) {
+ if (fmt_map[i].oss_fmt == fmt) {
+ GstStructure *s;
+
+ s = gst_structure_empty_new (fmt_map[i].name);
+ if (fmt_map[i].width != 0 && fmt_map[i].depth != 0) {
+ gst_structure_set (s, "width", G_TYPE_INT, fmt_map[i].width,
+ "depth", G_TYPE_INT, fmt_map[i].depth, "endianness", G_TYPE_INT,
+ fmt_map[i].endianness, "signed", G_TYPE_BOOLEAN,
+ fmt_map[i].signedness, NULL);
+ }
+ gst_caps_append_structure (caps, s);
+ return TRUE;
+ }
+ }
+ return FALSE;
+}
+
+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 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 ();
+
+ 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].oss_fmt, 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].oss_fmt, 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;
+}
+
+static gint
+gst_oss4_audio_ilog2 (gint x)
+{
+ /* well... hacker's delight explains... */
+ x = x | (x >> 1);
+ x = x | (x >> 2);
+ x = x | (x >> 4);
+ x = x | (x >> 8);
+ x = x | (x >> 16);
+ x = x - ((x >> 1) & 0x55555555);
+ x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
+ x = (x + (x >> 4)) & 0x0f0f0f0f;
+ x = x + (x >> 8);
+ x = x + (x >> 16);
+ return (x & 0x0000003f) - 1;
+}
+
+/* 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, fragsize;
+
+ 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;
+ }
+
+ /* CHECKME: maybe we should just leave the fragsize alone? (tpm) */
+ fragsize = gst_oss4_audio_ilog2 (spec->segsize);
+ fragsize = ((spec->segtotal & 0x7fff) << 16) | fragsize;
+ GST_DEBUG_OBJECT (obj, "setting segsize: %d, segtotal: %d, value: %08x",
+ spec->segsize, spec->segtotal, fragsize);
+
+ /* we could also use the new SNDCTL_DSP_POLICY if there's something in
+ * particular we're trying to achieve here */
+ if (ioctl (fd, SNDCTL_DSP_SETFRAGMENT, &fragsize) == -1)
+ goto set_fragsize_failed;
+
+ 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;
+ spec->segtotal = info.fragstotal;
+
+ spec->bytes_per_sample = (spec->width / 8) * spec->channels;
+
+ GST_DEBUG_OBJECT (obj, "got segsize: %d, segtotal: %d, value: %08x",
+ spec->segsize, spec->segtotal, 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;
+ }
+set_fragsize_failed:
+ {
+ GST_ELEMENT_ERROR (obj, RESOURCE, SETTINGS, (NULL),
+ ("DSP_SETFRAGMENT(%d) failed: %s", fragsize, 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);
+#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)
--- /dev/null
+/* 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.
+ */
+
+#ifndef GST_OSS4_AUDIO_H
+#define GST_OSS4_AUDIO_H_
+
+#include <gst/gst.h>
+#include <gst/audio/gstringbuffer.h>
+
+/* This is the minimum version we require */
+#define GST_MIN_OSS4_VERSION 0x040003
+
+int gst_oss4_audio_get_version (GstObject * obj, int fd);
+
+gboolean gst_oss4_audio_check_version (GstObject * obj, int fd);
+
+GstCaps * gst_oss4_audio_probe_caps (GstObject * obj, int fd);
+
+gboolean gst_oss4_audio_set_format (GstObject * obj, int fd, GstRingBufferSpec * spec);
+
+GstCaps * gst_oss4_audio_get_template_caps (void);
+
+gchar * gst_oss4_audio_find_device (GstObject * oss);
+
+#endif /* GST_OSS4_AUDIO_H */
+
+
--- /dev/null
+/* GStreamer OSS4 mixer enumeration control
+ * 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.
+ */
+
+/* An 'enum' in gnome-volume-control / GstMixer is represented by a
+ * GstMixerOptions object
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-mixer.h"
+#include "oss4-mixer-enum.h"
+#include "oss4-soundcard.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss4mixer_debug);
+#define GST_CAT_DEFAULT oss4mixer_debug
+
+static GList *gst_oss4_mixer_enum_get_values (GstMixerOptions * options);
+
+/* GstMixerTrack is a plain GObject, so let's just use the GLib macro here */
+G_DEFINE_TYPE (GstOss4MixerEnum, gst_oss4_mixer_enum, GST_TYPE_MIXER_OPTIONS);
+
+static void
+gst_oss4_mixer_enum_init (GstOss4MixerEnum * e)
+{
+ e->need_update = TRUE;
+}
+
+static void
+gst_oss4_mixer_enum_dispose (GObject * obj)
+{
+ GstMixerOptions *options = GST_MIXER_OPTIONS (obj);
+
+ /* our list is a flat list with constant strings, but the GstMixerOptions
+ * dispose will try to g_free the contained strings, so clean up the list
+ * before chaining up to GstMixerOptions */
+ g_list_free (options->values);
+ options->values = NULL;
+
+ G_OBJECT_CLASS (gst_oss4_mixer_enum_parent_class)->dispose (obj);
+}
+
+static void
+gst_oss4_mixer_enum_class_init (GstOss4MixerEnumClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstMixerOptionsClass *mixeroptions_class = (GstMixerOptionsClass *) klass;
+
+ gobject_class->dispose = gst_oss4_mixer_enum_dispose;
+ mixeroptions_class->get_values = gst_oss4_mixer_enum_get_values;
+}
+
+static GList *
+gst_oss4_mixer_enum_get_values_locked (GstMixerOptions * options)
+{
+ GstOss4MixerEnum *e = GST_OSS4_MIXER_ENUM_CAST (options);
+ GList *oldlist, *list = NULL;
+ int i;
+
+ /* if current list of values is empty, update/re-check in any case */
+ if (!e->need_update && options->values != NULL)
+ return options->values;
+
+ GST_LOG_OBJECT (e, "updating available values for %s", e->mc->mixext.extname);
+
+ for (i = 0; i < e->mc->mixext.maxvalue; ++i) {
+ if (MIXEXT_ENUM_IS_AVAILABLE (e->mc->mixext, i)) {
+ const gchar *s;
+
+ s = g_quark_to_string (e->mc->enum_vals[i]);
+ GST_LOG_OBJECT (e, "option '%s' is available", s);
+ list = g_list_prepend (list, (gpointer) s);
+ } else {
+ GST_LOG_OBJECT (e, "option '%s' is currently not available");
+ }
+ }
+
+ list = g_list_reverse (list);
+
+ /* this is not thread-safe, but then the entire GstMixer API isn't really,
+ * since we return foo->list and not a copy and don't take any locks, so
+ * not much we can do here but pray; we're usually either called from _new()
+ * or from within _get_values() though, so it should be okay. We could use
+ * atomic ops here, but I'm not sure how much more that really buys us.*/
+ oldlist = options->values; /* keep window small */
+ options->values = list;
+ g_list_free (oldlist);
+
+ e->need_update = FALSE;
+
+ return options->values;
+}
+
+static GList *
+gst_oss4_mixer_enum_get_values (GstMixerOptions * options)
+{
+ GstOss4MixerEnum *e = GST_OSS4_MIXER_ENUM (options);
+ GList *list;
+
+ /* we take the lock here mostly to serialise ioctls with the watch thread */
+ GST_OBJECT_LOCK (e->mixer);
+
+ list = gst_oss4_mixer_enum_get_values_locked (options);
+
+ GST_OBJECT_UNLOCK (e->mixer);
+
+ return list;
+}
+
+static const gchar *
+gst_oss4_mixer_enum_get_current_value (GstOss4MixerEnum * e)
+{
+ const gchar *cur_val = NULL;
+
+ if (e->mc->enum_vals != NULL && e->mc->last_val < e->mc->mixext.maxvalue) {
+ cur_val = g_quark_to_string (e->mc->enum_vals[e->mc->last_val]);
+ }
+
+ return cur_val;
+}
+
+static gboolean
+gst_oss4_mixer_enum_update_current (GstOss4MixerEnum * e)
+{
+ int cur = -1;
+
+ if (!gst_oss4_mixer_get_control_val (e->mixer, e->mc, &cur))
+ return FALSE;
+
+ if (cur < 0 || cur >= e->mc->mixext.maxvalue) {
+ GST_WARNING_OBJECT (e, "read value %d out of bounds [0-%d]", cur,
+ e->mc->mixext.maxvalue - 1);
+ e->mc->last_val = 0;
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+gboolean
+gst_oss4_mixer_enum_set_option (GstOss4MixerEnum * e, const gchar * value)
+{
+ GQuark q;
+ int i;
+
+ q = g_quark_try_string (value);
+ if (q == 0) {
+ GST_WARNING_OBJECT (e, "unknown option '%s'", value);
+ return FALSE;
+ }
+
+ for (i = 0; i < e->mc->mixext.maxvalue; ++i) {
+ if (q == e->mc->enum_vals[i])
+ break;
+ }
+
+ if (i >= e->mc->mixext.maxvalue) {
+ GST_WARNING_OBJECT (e, "option '%s' is not valid for this control", value);
+ return FALSE;
+ }
+
+ GST_LOG_OBJECT (e, "option '%s' = %d", value, i);
+
+ if (!MIXEXT_ENUM_IS_AVAILABLE (e->mc->mixext, i)) {
+ GST_WARNING_OBJECT (e, "option '%s' is not selectable currently", value);
+ return FALSE;
+ }
+
+ if (!gst_oss4_mixer_set_control_val (e->mixer, e->mc, i)) {
+ GST_WARNING_OBJECT (e, "could not set option '%s' (%d)", value, i);
+ return FALSE;
+ }
+
+ /* and re-read current value with sanity checks (or could just assign here) */
+ gst_oss4_mixer_enum_update_current (e);
+
+ return TRUE;
+}
+
+const gchar *
+gst_oss4_mixer_enum_get_option (GstOss4MixerEnum * e)
+{
+ const gchar *cur_str = NULL;
+
+ if (!gst_oss4_mixer_enum_update_current (e)) {
+ GST_WARNING_OBJECT (e, "failed to read current value");
+ return NULL;
+ }
+
+ cur_str = gst_oss4_mixer_enum_get_current_value (e);
+ GST_LOG_OBJECT (e, "%s (%d)", GST_STR_NULL (cur_str), e->mc->last_val);
+ return cur_str;
+}
+
+GstMixerTrack *
+gst_oss4_mixer_enum_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc)
+{
+ GstOss4MixerEnum *e;
+ GstMixerTrack *track;
+
+ e = g_object_new (GST_TYPE_OSS4_MIXER_ENUM, "untranslated-label",
+ mc->mixext.extname, NULL);
+ e->mixer = mixer;
+ e->mc = mc;
+
+ track = GST_MIXER_TRACK (e);
+
+ /* caller will set track->label and track->flags */
+
+ track->num_channels = 0;
+ track->min_volume = 0;
+ track->max_volume = 0;
+
+ (void) gst_oss4_mixer_enum_get_values_locked (GST_MIXER_OPTIONS (track));
+
+ if (!gst_oss4_mixer_enum_update_current (e)) {
+ GST_WARNING_OBJECT (track, "failed to read current value, returning NULL");
+ g_object_unref (track);
+ track = NULL;
+ }
+
+ GST_LOG_OBJECT (e, "current value: %d (%s)", e->mc->last_val,
+ gst_oss4_mixer_enum_get_current_value (e));
+
+ return track;
+}
+
+/* This is called from the watch thread */
+void
+gst_oss4_mixer_enum_process_change_unlocked (GstMixerTrack * track)
+{
+ GstOss4MixerEnum *e = GST_OSS4_MIXER_ENUM_CAST (track);
+
+ gchar *cur;
+
+ if (!e->mc->changed && !e->mc->list_changed)
+ return;
+
+ if (e->mc->list_changed) {
+ gst_mixer_options_list_changed (GST_MIXER (e->mixer),
+ GST_MIXER_OPTIONS (e));
+ }
+
+ GST_OBJECT_LOCK (e->mixer);
+ cur = (gchar *) gst_oss4_mixer_enum_get_current_value (e);
+ GST_OBJECT_UNLOCK (e->mixer);
+
+ gst_mixer_option_changed (GST_MIXER (e->mixer), GST_MIXER_OPTIONS (e), cur);
+}
--- /dev/null
+/* GStreamer OSS4 mixer on/off enum control
+ * 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.
+ */
+
+#ifndef GST_OSS4_MIXER_ENUM_H
+#define GST_OSS4_MIXER_ENUM_H
+
+#include <gst/gst.h>
+#include <gst/interfaces/mixer.h>
+
+#include "oss4-mixer.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_MIXER_ENUM (gst_oss4_mixer_enum_get_type())
+#define GST_OSS4_MIXER_ENUM(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_MIXER_ENUM,GstOss4MixerEnum))
+#define GST_OSS4_MIXER_ENUM_CAST(obj) ((GstOss4MixerEnum *)(obj))
+#define GST_OSS4_MIXER_ENUM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_MIXER_ENUM,GstOss4MixerEnumClass))
+#define GST_IS_OSS4_MIXER_ENUM(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_MIXER_ENUM))
+#define GST_IS_OSS4_MIXER_ENUM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_MIXER_ENUM))
+
+typedef struct _GstOss4MixerEnum GstOss4MixerEnum;
+typedef struct _GstOss4MixerEnumClass GstOss4MixerEnumClass;
+
+struct _GstOss4MixerEnum {
+ GstMixerOptions mixer_option;
+
+ GstOss4MixerControl * mc;
+ GstOss4Mixer * mixer; /* the mixer we belong to (no ref taken) */
+
+ gboolean need_update;
+};
+
+struct _GstOss4MixerEnumClass {
+ GstMixerOptionsClass mixer_option_class;
+};
+
+GType gst_oss4_mixer_enum_get_type (void);
+
+gboolean gst_oss4_mixer_enum_set_option (GstOss4MixerEnum * e, const gchar * value);
+
+const gchar * gst_oss4_mixer_enum_get_option (GstOss4MixerEnum * e);
+
+GstMixerTrack * gst_oss4_mixer_enum_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc);
+
+void gst_oss4_mixer_enum_process_change_unlocked (GstMixerTrack * track);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_MIXER_ENUM_H */
+
+
--- /dev/null
+/* GStreamer OSS4 mixer slider control
+ * 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.
+ */
+
+/* A 'slider' in gnome-volume-control / GstMixer is represented by a
+ * GstMixerTrack with one or more channels.
+ *
+ * A slider should be either flagged as INPUT or OUTPUT (mostly because of
+ * gnome-volume-control being littered with g_asserts for everything it doesn't
+ * expect).
+ *
+ * From mixertrack.h:
+ * "Input tracks can have 'recording' enabled, which means that any input will
+ * be hearable into the speakers that are attached to the output. Mute is
+ * obvious."
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-mixer-slider.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss4mixer_debug);
+#define GST_CAT_DEFAULT oss4mixer_debug
+
+/* GstMixerTrack is a plain GObject, so let's just use the GLib macro here */
+G_DEFINE_TYPE (GstOss4MixerSlider, gst_oss4_mixer_slider, GST_TYPE_MIXER_TRACK);
+
+static void
+gst_oss4_mixer_slider_class_init (GstOss4MixerSliderClass * klass)
+{
+ /* nothing to do here */
+}
+
+static void
+gst_oss4_mixer_slider_init (GstOss4MixerSlider * s)
+{
+ /* nothing to do here */
+}
+
+static int
+gst_oss4_mixer_slider_pack_volume (GstOss4MixerSlider * s, const gint * volumes)
+{
+ int val = 0;
+
+ switch (s->mc->mixext.type) {
+ case MIXT_MONOSLIDER:
+ case MIXT_MONOSLIDER16:
+ case MIXT_SLIDER:
+ val = volumes[0];
+ break;
+ case MIXT_STEREOSLIDER:
+ val = ((volumes[1] & 0xff) << 8) | (volumes[0] & 0xff);
+ break;
+ case MIXT_STEREOSLIDER16:
+ val = ((volumes[1] & 0xffff) << 16) | (volumes[0] & 0xffff);
+ break;
+ default:
+ g_return_val_if_reached (0);
+ }
+ return val;
+}
+
+static void
+gst_oss4_mixer_slider_unpack_volume (GstOss4MixerSlider * s, int v,
+ gint * volumes)
+{
+ guint32 val; /* use uint so bitshifting the highest bit works right */
+
+ val = (guint32) v;
+ switch (s->mc->mixext.type) {
+ case MIXT_MONOSLIDER:
+ case MIXT_MONOSLIDER16:
+ case MIXT_SLIDER:
+ volumes[0] = val;
+ break;
+ case MIXT_STEREOSLIDER:
+ volumes[0] = (val & 0x00ff);
+ volumes[1] = (val & 0xff00) >> 8;
+ break;
+ case MIXT_STEREOSLIDER16:
+ volumes[0] = (val & 0x0000ffff);
+ volumes[1] = (val & 0xffff0000) >> 16;
+ break;
+ default:
+ g_return_if_reached ();
+ }
+}
+
+gboolean
+gst_oss4_mixer_slider_get_volume (GstOss4MixerSlider * s, gint * volumes)
+{
+ GstMixerTrack *track = GST_MIXER_TRACK (s);
+ int v = 0;
+
+ /* if we're supposed to be muted, and don't have an actual mute control
+ * (ie. 'simulate' the mute), then just return the volume as saved, not
+ * the actually set volume which is most likely 0 */
+ if (GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_MUTE) && !s->mc->mute) {
+ volumes[0] = s->volumes[0];
+ if (track->num_channels == 2)
+ volumes[1] = s->volumes[1];
+ return TRUE;
+ }
+
+ if (!gst_oss4_mixer_get_control_val (s->mixer, s->mc, &v))
+ return FALSE;
+
+ gst_oss4_mixer_slider_unpack_volume (s, v, volumes);
+
+ if (track->num_channels > 1) {
+ GST_LOG_OBJECT (s, "volume: left=%d, right=%d", volumes[0], volumes[1]);
+ } else {
+ GST_LOG_OBJECT (s, "volume: mono=%d", volumes[0]);
+ }
+
+ return TRUE;
+}
+
+gboolean
+gst_oss4_mixer_slider_set_volume (GstOss4MixerSlider * s, const gint * volumes)
+{
+ GstMixerTrack *track = GST_MIXER_TRACK (s);
+ int val = 0;
+
+ /* if we're supposed to be muted, and are 'simulating' the mute because
+ * we don't have a mute control, don't actually change the volume, just
+ * save it as the new desired volume for later when we get unmuted again */
+ if (GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_MUTE) && !s->mc->mute)
+ goto done;
+
+ val = gst_oss4_mixer_slider_pack_volume (s, volumes);
+
+ if (track->num_channels > 1) {
+ GST_LOG_OBJECT (s, "left=%d, right=%d", volumes[0], volumes[1]);
+ } else {
+ GST_LOG_OBJECT (s, "mono=%d", volumes[0]);
+ }
+
+ if (!gst_oss4_mixer_set_control_val (s->mixer, s->mc, val))
+ return FALSE;
+
+done:
+
+ s->volumes[0] = volumes[0];
+ if (track->num_channels == 2)
+ s->volumes[1] = volumes[1];
+
+ return TRUE;
+}
+
+gboolean
+gst_oss4_mixer_slider_set_record (GstOss4MixerSlider * s, gboolean record)
+{
+ /* There doesn't seem to be a way to do this using the OSS4 mixer API, so
+ * just do nothing here for now. */
+ return FALSE;
+}
+
+gboolean
+gst_oss4_mixer_slider_set_mute (GstOss4MixerSlider * s, gboolean mute)
+{
+ GstMixerTrack *track = GST_MIXER_TRACK (s);
+ gboolean ret;
+
+ /* if we don't have a mute control, simulate mute (which is a bit broken,
+ * since we can't differentiate between capture/playback volume etc., so
+ * we just assume that setting the volume to 0 would be the same as muting
+ * this control) */
+ if (s->mc->mute == NULL) {
+ int volume;
+
+ if (mute) {
+ volume = 0;
+ } else {
+ volume = gst_oss4_mixer_slider_pack_volume (s, s->volumes);
+ }
+ ret = gst_oss4_mixer_set_control_val (s->mixer, s->mc, volume);
+ } else {
+ ret = gst_oss4_mixer_set_control_val (s->mixer, s->mc->mute, !!mute);
+ }
+
+ if (mute) {
+ track->flags |= GST_MIXER_TRACK_MUTE;
+ } else {
+ track->flags &= ~GST_MIXER_TRACK_MUTE;
+ }
+
+ return FALSE;
+}
+
+GstMixerTrack *
+gst_oss4_mixer_slider_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc)
+{
+ GstOss4MixerSlider *s;
+ GstMixerTrack *track;
+ gint volumes[2] = { 0, };
+
+ s = g_object_new (GST_TYPE_OSS4_MIXER_SLIDER, "untranslated-label",
+ mc->mixext.extname, NULL);
+
+ track = GST_MIXER_TRACK (s);
+
+ /* caller will set track->label and track->flags */
+
+ s->mc = mc;
+ s->mixer = mixer;
+
+ /* we don't do value scaling but just present a scale of 0-maxvalue */
+ track->min_volume = 0;
+ track->max_volume = mc->mixext.maxvalue;
+
+ switch (mc->mixext.type) {
+ case MIXT_MONOSLIDER:
+ case MIXT_MONOSLIDER16:
+ case MIXT_SLIDER:
+ track->num_channels = 1;
+ break;
+ case MIXT_STEREOSLIDER:
+ case MIXT_STEREOSLIDER16:
+ track->num_channels = 2;
+ break;
+ default:
+ g_return_val_if_reached (NULL);
+ }
+
+ GST_LOG_OBJECT (track, "min=%d, max=%d, channels=%d", track->min_volume,
+ track->max_volume, track->num_channels);
+
+ if (!gst_oss4_mixer_slider_get_volume (s, volumes)) {
+ GST_WARNING_OBJECT (track, "failed to read volume, returning NULL");
+ g_object_unref (track);
+ track = NULL;
+ }
+
+ return track;
+}
+
+/* This is called from the watch thread */
+void
+gst_oss4_mixer_slider_process_change_unlocked (GstMixerTrack * track)
+{
+ GstOss4MixerSlider *s = GST_OSS4_MIXER_SLIDER_CAST (track);
+
+ if (s->mc->mute != NULL && s->mc->mute->changed) {
+ gst_mixer_mute_toggled (GST_MIXER (s->mixer), track,
+ !!s->mc->mute->last_val);
+ } else {
+ /* nothing to do here, since we don't/can't easily implement the record
+ * flag */
+ }
+
+ if (s->mc->changed) {
+ gint volumes[2] = { 0, 0 };
+
+ gst_oss4_mixer_slider_unpack_volume (s, s->mc->last_val, volumes);
+
+ /* if we 'simulate' the mute, update flag when the volume changes */
+ if (s->mc->mute == NULL) {
+ if (volumes[0] == 0 && volumes[1] == 0) {
+ track->flags |= GST_MIXER_TRACK_MUTE;
+ } else {
+ track->flags &= ~GST_MIXER_TRACK_MUTE;
+ }
+ }
+
+ gst_mixer_volume_changed (GST_MIXER (s->mixer), track, volumes);
+ }
+}
--- /dev/null
+/* GStreamer OSS4 mixer slider control
+ * 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.
+ */
+
+#ifndef GST_OSS4_MIXER_SLIDER_H
+#define GST_OSS4_MIXER_SLIDER_H
+
+#include <gst/gst.h>
+#include <gst/interfaces/mixer.h>
+
+#include "oss4-mixer.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_MIXER_SLIDER (gst_oss4_mixer_slider_get_type())
+#define GST_OSS4_MIXER_SLIDER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_MIXER_SLIDER,GstOss4MixerSlider))
+#define GST_OSS4_MIXER_SLIDER_CAST(obj) ((GstOss4MixerSlider *)(obj))
+#define GST_OSS4_MIXER_SLIDER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_MIXER_SLIDER,GstOss4MixerSliderClass))
+#define GST_IS_OSS4_MIXER_SLIDER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_MIXER_SLIDER))
+#define GST_IS_OSS4_MIXER_SLIDER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_MIXER_SLIDER))
+
+typedef struct _GstOss4MixerSlider GstOss4MixerSlider;
+typedef struct _GstOss4MixerSliderClass GstOss4MixerSliderClass;
+
+struct _GstOss4MixerSlider {
+ GstMixerTrack mixer_track;
+
+ GstOss4MixerControl * mc;
+ GstOss4Mixer * mixer; /* the mixer we belong to (no ref taken) */
+ gint volumes[2]; /* left/mono, right */
+};
+
+struct _GstOss4MixerSliderClass {
+ GstMixerTrackClass mixer_track_class;
+};
+
+GType gst_oss4_mixer_slider_get_type (void);
+
+GstMixerTrack * gst_oss4_mixer_slider_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc);
+
+gboolean gst_oss4_mixer_slider_get_volume (GstOss4MixerSlider * s, gint * volumes);
+
+gboolean gst_oss4_mixer_slider_set_volume (GstOss4MixerSlider * s, const gint * volumes);
+
+gboolean gst_oss4_mixer_slider_set_record (GstOss4MixerSlider * s, gboolean record);
+
+gboolean gst_oss4_mixer_slider_set_mute (GstOss4MixerSlider * s, gboolean mute);
+
+void gst_oss4_mixer_slider_process_change_unlocked (GstMixerTrack * track);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_MIXER_SLIDER_H */
+
+
--- /dev/null
+/* GStreamer OSS4 mixer on/off switch control
+ * 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.
+ */
+
+/* A simple ON/OFF 'switch' in gnome-volume-control / GstMixer is represented
+ * by a GstMixerTrack with no channels.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-mixer-switch.h"
+#include "oss4-soundcard.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss4mixer_debug);
+#define GST_CAT_DEFAULT oss4mixer_debug
+
+/* GstMixerTrack is a plain GObject, so let's just use the GLib macro here */
+G_DEFINE_TYPE (GstOss4MixerSwitch, gst_oss4_mixer_switch, GST_TYPE_MIXER_TRACK);
+
+static void
+gst_oss4_mixer_switch_class_init (GstOss4MixerSwitchClass * klass)
+{
+ /* nothing to do here */
+}
+
+static void
+gst_oss4_mixer_switch_init (GstOss4MixerSwitch * s)
+{
+ /* nothing to do here */
+}
+
+static GstMixerTrackFlags
+gst_oss4_mixer_switch_get_switch_flag (GstMixerTrack * track)
+{
+ if ((track->flags & GST_MIXER_TRACK_INPUT)) {
+ return GST_MIXER_TRACK_RECORD;
+ } else if ((track->flags & GST_MIXER_TRACK_OUTPUT)) {
+ return GST_MIXER_TRACK_MUTE;
+ } else {
+ GST_ERROR_OBJECT (track, "switch neither input nor output track!?");
+ }
+ return 0;
+}
+
+gboolean
+gst_oss4_mixer_switch_set (GstOss4MixerSwitch * s, gboolean enabled)
+{
+ GstMixerTrackFlags switch_flag;
+ GstMixerTrack *track;
+ int newval;
+
+ track = GST_MIXER_TRACK (s);
+ switch_flag = gst_oss4_mixer_switch_get_switch_flag (track);
+
+ newval = (enabled) ? 1 : 0;
+
+ if (!!newval == !!(track->flags & switch_flag)) {
+ GST_LOG_OBJECT (s, "switch is already %d, doing nothing", newval);
+ return TRUE;
+ }
+
+ if (!gst_oss4_mixer_set_control_val (s->mixer, s->mc, newval)) {
+ GST_WARNING_OBJECT (s, "could not set switch to %d", newval);
+ return FALSE;
+ }
+
+ if (newval) {
+ track->flags |= switch_flag;
+ } else {
+ track->flags &= ~switch_flag;
+ }
+
+ GST_LOG_OBJECT (s, "set switch to %d", newval);
+
+ return TRUE;
+}
+
+gboolean
+gst_oss4_mixer_switch_get (GstOss4MixerSwitch * s, gboolean * enabled)
+{
+ GstMixerTrackFlags switch_flag;
+ GstMixerTrack *track;
+ int val = -1;
+
+ track = GST_MIXER_TRACK (s);
+ switch_flag = gst_oss4_mixer_switch_get_switch_flag (track);
+
+ if (!gst_oss4_mixer_get_control_val (s->mixer, s->mc, &val) || val < 0) {
+ GST_WARNING_OBJECT (s, "could not get switch state");
+ return FALSE;
+ }
+
+ *enabled = (val != 0);
+
+ if (!!val != !!(track->flags & switch_flag)) {
+ GST_INFO_OBJECT (s, "updating inconsistent switch state to %d", !!val);
+ if (*enabled) {
+ track->flags |= switch_flag;
+ } else {
+ track->flags &= ~switch_flag;
+ }
+ }
+
+
+ return TRUE;
+}
+
+GstMixerTrack *
+gst_oss4_mixer_switch_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc)
+{
+ GstOss4MixerSwitch *s;
+ GstMixerTrack *track;
+ int cur = -1;
+
+ s = g_object_new (GST_TYPE_OSS4_MIXER_SWITCH, "untranslated-label",
+ mc->mixext.extname, NULL);
+
+ s->mixer = mixer;
+ s->mc = mc;
+
+ track = GST_MIXER_TRACK (s);
+
+ /* caller will set track->label and track->flags */
+
+ track->num_channels = 0;
+ track->min_volume = 0;
+ track->max_volume = 0;
+
+ if (!gst_oss4_mixer_get_control_val (s->mixer, s->mc, &cur) || cur < 0)
+ return NULL;
+
+ return track;
+}
+
+/* This is called from the watch thread */
+void
+gst_oss4_mixer_switch_process_change_unlocked (GstMixerTrack * track)
+{
+ GstOss4MixerSwitch *s = GST_OSS4_MIXER_SWITCH_CAST (track);
+
+ if (!s->mc->changed)
+ return;
+
+ if ((track->flags & GST_MIXER_TRACK_INPUT)) {
+ gst_mixer_record_toggled (GST_MIXER (s->mixer), track, !!s->mc->last_val);
+ } else {
+ gst_mixer_mute_toggled (GST_MIXER (s->mixer), track, !!s->mc->last_val);
+ }
+}
--- /dev/null
+/* GStreamer OSS4 mixer on/off switch control
+ * 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.
+ */
+
+#ifndef GST_OSS4_MIXER_SWITCH_H
+#define GST_OSS4_MIXER_SWITCH_H
+
+#include <gst/gst.h>
+#include <gst/interfaces/mixer.h>
+
+#include "oss4-mixer.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_MIXER_SWITCH (gst_oss4_mixer_switch_get_type())
+#define GST_OSS4_MIXER_SWITCH(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_MIXER_SWITCH,GstOss4MixerSwitch))
+#define GST_OSS4_MIXER_SWITCH_CAST(obj) ((GstOss4MixerSwitch *)(obj))
+#define GST_OSS4_MIXER_SWITCH_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_MIXER_SWITCH,GstOss4MixerSwitchClass))
+#define GST_IS_OSS4_MIXER_SWITCH(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_MIXER_SWITCH))
+#define GST_IS_OSS4_MIXER_SWITCH_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_MIXER_SWITCH))
+
+typedef struct _GstOss4MixerSwitch GstOss4MixerSwitch;
+typedef struct _GstOss4MixerSwitchClass GstOss4MixerSwitchClass;
+
+struct _GstOss4MixerSwitch {
+ GstMixerTrack mixer_track;
+
+ GstOss4MixerControl * mc;
+ GstOss4Mixer * mixer; /* the mixer we belong to (no ref taken) */
+};
+
+struct _GstOss4MixerSwitchClass {
+ GstMixerTrackClass mixer_track_class;
+};
+
+GType gst_oss4_mixer_switch_get_type (void);
+
+gboolean gst_oss4_mixer_switch_set (GstOss4MixerSwitch * s, gboolean enabled);
+
+gboolean gst_oss4_mixer_switch_get (GstOss4MixerSwitch * s, gboolean * enabled);
+
+GstMixerTrack * gst_oss4_mixer_switch_new (GstOss4Mixer * mixer, GstOss4MixerControl * mc);
+
+void gst_oss4_mixer_switch_process_change_unlocked (GstMixerTrack * track);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_MIXER_SWITCH_H */
+
+
--- /dev/null
+/* GStreamer OSS4 mixer implementation
+ * 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 mixer library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-oss4mixer
+ * @short_description: element to control sound input and output levels with OSS4
+ *
+ * <refsect2>
+ * <para>
+ * This element lets you adjust sound input and output levels with the
+ * Open Sound System (OSS) version 4. It supports the GstMixer interface, which
+ * can be used to obtain a list of available mixer tracks. Set the mixer
+ * element to READY state before using the GstMixer interface on it.
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * oss4mixer can't be used in a sensible way in gst-launch.
+ * </para>
+ * </refsect2>
+ *
+ * Since: 0.10.7
+ */
+
+/* Note: ioctl calls on the same open mixer device are serialised via
+ * the object lock to make sure we don't do concurrent ioctls from two
+ * different threads (e.g. app thread and mixer watch thread), since that
+ * will probably confuse OSS. */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+
+#include <gst/interfaces/mixer.h>
+#include <gst/gst-i18n-plugin.h>
+
+#include <glib/gprintf.h>
+
+#define NO_LEGACY_MIXER
+
+#include "oss4-audio.h"
+#include "oss4-mixer.h"
+#include "oss4-mixer-enum.h"
+#include "oss4-mixer-slider.h"
+#include "oss4-mixer-switch.h"
+#include "oss4-property-probe.h"
+#include "oss4-soundcard.h"
+
+#define GST_OSS4_MIXER_WATCH_INTERVAL 500 /* in millisecs, ie. 0.5s */
+
+GST_DEBUG_CATEGORY_EXTERN (oss4mixer_debug);
+#define GST_CAT_DEFAULT oss4mixer_debug
+
+#define DEFAULT_DEVICE NULL
+#define DEFAULT_DEVICE_NAME NULL
+
+enum
+{
+ PROP_0,
+ PROP_DEVICE,
+ PROP_DEVICE_NAME
+};
+
+static void gst_oss4_mixer_init_interfaces (GType type);
+
+GST_BOILERPLATE_FULL (GstOss4Mixer, gst_oss4_mixer, GstElement,
+ GST_TYPE_ELEMENT, gst_oss4_mixer_init_interfaces);
+
+static GstStateChangeReturn gst_oss4_mixer_change_state (GstElement *
+ element, GstStateChange transition);
+
+static void gst_oss4_mixer_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_oss4_mixer_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void gst_oss4_mixer_finalize (GObject * object);
+
+static gboolean gst_oss4_mixer_open (GstOss4Mixer * mixer,
+ gboolean silent_errors);
+static void gst_oss4_mixer_close (GstOss4Mixer * mixer);
+
+static gboolean gst_oss4_mixer_enum_control_update_enum_list (GstOss4Mixer * m,
+ GstOss4MixerControl * mc);
+
+#ifndef GST_DISABLE_DEBUG
+static const gchar *mixer_ext_type_get_name (gint type);
+static const gchar *mixer_ext_flags_get_string (gint flags);
+#endif
+
+static void
+gst_oss4_mixer_base_init (gpointer klass)
+{
+ gst_element_class_set_details_simple (GST_ELEMENT_CLASS (klass),
+ "OSS v4 Audio Mixer", "Generic/Audio",
+ "Control sound input and output levels with OSS4",
+ "Tim-Philipp Müller <tim centricular net>");
+}
+
+static void
+gst_oss4_mixer_class_init (GstOss4MixerClass * klass)
+{
+ GstElementClass *element_class;
+ GObjectClass *gobject_class;
+
+ element_class = (GstElementClass *) klass;
+ gobject_class = (GObjectClass *) klass;
+
+ gobject_class->finalize = gst_oss4_mixer_finalize;
+ gobject_class->set_property = gst_oss4_mixer_set_property;
+ gobject_class->get_property = gst_oss4_mixer_get_property;
+
+ /**
+ * GstOss4Mixer:device
+ *
+ * OSS4 mixer device (e.g. /dev/oss/hdaudio0/mix0 or /dev/mixerN)
+ *
+ **/
+ g_object_class_install_property (gobject_class, PROP_DEVICE,
+ g_param_spec_string ("device", "Device",
+ "OSS mixer device (e.g. /dev/oss/hdaudio0/mix0 or /dev/mixerN) "
+ "(NULL = use first mixer device found)", DEFAULT_DEVICE,
+ G_PARAM_READWRITE));
+
+ /**
+ * GstOss4Mixer:device-name
+ *
+ * Human-readable name of the sound device. May be NULL if the device is
+ * not open (ie. when the mixer is in NULL state)
+ *
+ **/
+ g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
+ g_param_spec_string ("device-name", "Device name",
+ "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
+ G_PARAM_READABLE));
+
+ element_class->change_state = GST_DEBUG_FUNCPTR (gst_oss4_mixer_change_state);
+}
+
+static void
+gst_oss4_mixer_finalize (GObject * obj)
+{
+ GstOss4Mixer *mixer = GST_OSS4_MIXER (obj);
+
+ g_free (mixer->device);
+
+ G_OBJECT_CLASS (parent_class)->finalize (obj);
+}
+
+static void
+gst_oss4_mixer_reset (GstOss4Mixer * mixer)
+{
+ mixer->fd = -1;
+ mixer->need_update = TRUE;
+ memset (&mixer->last_mixext, 0, sizeof (oss_mixext));
+}
+
+static void
+gst_oss4_mixer_init (GstOss4Mixer * mixer, GstOss4MixerClass * g_class)
+{
+ mixer->device = g_strdup (DEFAULT_DEVICE);
+ mixer->device_name = NULL;
+
+ gst_oss4_mixer_reset (mixer);
+}
+
+static void
+gst_oss4_mixer_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstOss4Mixer *mixer = GST_OSS4_MIXER (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ GST_OBJECT_LOCK (mixer);
+ if (!GST_OSS4_MIXER_IS_OPEN (mixer)) {
+ g_free (mixer->device);
+ mixer->device = g_value_dup_string (value);
+ /* unset any cached device-name */
+ g_free (mixer->device_name);
+ mixer->device_name = NULL;
+ } else {
+ g_warning ("%s: can't change \"device\" property while mixer is open",
+ GST_OBJECT_NAME (mixer));
+ }
+ GST_OBJECT_UNLOCK (mixer);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_oss4_mixer_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstOss4Mixer *mixer = GST_OSS4_MIXER (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ GST_OBJECT_LOCK (mixer);
+ g_value_set_string (value, mixer->device);
+ GST_OBJECT_UNLOCK (mixer);
+ break;
+ case PROP_DEVICE_NAME:
+ GST_OBJECT_LOCK (mixer);
+ /* If device is set, try to retrieve the name even if we're not open */
+ if (mixer->fd == -1 && mixer->device != NULL) {
+ if (gst_oss4_mixer_open (mixer, TRUE)) {
+ g_value_set_string (value, mixer->device_name);
+ gst_oss4_mixer_close (mixer);
+ } else {
+ g_value_set_string (value, mixer->device_name);
+ }
+ } else {
+ g_value_set_string (value, mixer->device_name);
+ }
+ GST_OBJECT_UNLOCK (mixer);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static gboolean
+gst_oss4_mixer_open (GstOss4Mixer * mixer, gboolean silent_errors)
+{
+ struct oss_mixerinfo mi = { 0, };
+ gchar *device;
+
+ g_return_val_if_fail (!GST_OSS4_MIXER_IS_OPEN (mixer), FALSE);
+
+ if (mixer->device)
+ device = g_strdup (mixer->device);
+ else
+ device = gst_oss4_audio_find_device (GST_OBJECT_CAST (mixer));
+
+ /* desperate times, desperate measures */
+ if (device == NULL)
+ device = g_strdup ("/dev/mixer");
+
+ GST_INFO_OBJECT (mixer, "Trying to open OSS4 mixer device '%s'", device);
+
+ mixer->fd = open (device, O_RDWR, 0);
+ if (mixer->fd < 0)
+ goto open_failed;
+
+ /* Make sure it's OSS4. If it's old OSS, let the old ossmixer handle it */
+ if (!gst_oss4_audio_check_version (GST_OBJECT (mixer), mixer->fd))
+ goto legacy_oss;
+
+ GST_INFO_OBJECT (mixer, "Opened mixer device '%s', which is mixer %d",
+ device, mi.dev);
+
+ /* Get device name for currently open fd */
+ mi.dev = -1;
+ if (ioctl (mixer->fd, SNDCTL_MIXERINFO, &mi) == 0) {
+ mixer->modify_counter = mi.modify_counter;
+ if (mi.name[0] != '\0') {
+ mixer->device_name = g_strdup (mi.name);
+ }
+ } else {
+ mixer->modify_counter = 0;
+ }
+
+ if (mixer->device_name == NULL) {
+ mixer->device_name = g_strdup ("Unknown");
+ }
+ GST_INFO_OBJECT (mixer, "device name = '%s'", mixer->device_name);
+
+ mixer->open_device = device;
+
+ return TRUE;
+
+ /* ERRORS */
+open_failed:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (mixer, RESOURCE, OPEN_READ_WRITE,
+ (_("Could not open audio device for mixer control handling.")),
+ GST_ERROR_SYSTEM);
+ }
+ g_free (device);
+ return FALSE;
+ }
+legacy_oss:
+ {
+ gst_oss4_mixer_close (mixer);
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (mixer, RESOURCE, OPEN_READ_WRITE,
+ (_("Could not open audio device for mixer control handling."
+ "This version of the Open Sound System is not supported by this "
+ "element.")), ("Try the 'ossmixer' element instead"));
+ }
+ g_free (device);
+ return FALSE;
+ }
+}
+
+static void
+gst_oss4_mixer_control_free (GstOss4MixerControl * mc)
+{
+ g_list_free (mc->children);
+ g_list_free (mc->mute_group);
+ g_free (mc->enum_vals);
+ memset (mc, 0, sizeof (GstOss4MixerControl));
+ g_free (mc);
+}
+
+static void
+gst_oss4_mixer_free_tracks (GstOss4Mixer * mixer)
+{
+ g_list_foreach (mixer->tracks, (GFunc) g_object_unref, NULL);
+ g_list_free (mixer->tracks);
+ mixer->tracks = NULL;
+
+ g_list_foreach (mixer->controls, (GFunc) gst_oss4_mixer_control_free, NULL);
+ g_list_free (mixer->controls);
+ mixer->controls = NULL;
+}
+
+static void
+gst_oss4_mixer_close (GstOss4Mixer * mixer)
+{
+ g_free (mixer->device_name);
+ mixer->device_name = NULL;
+
+ g_free (mixer->open_device);
+ mixer->open_device = NULL;
+
+ gst_oss4_mixer_free_tracks (mixer);
+
+ if (mixer->fd != -1) {
+ close (mixer->fd);
+ mixer->fd = -1;
+ }
+
+ gst_oss4_mixer_reset (mixer);
+}
+
+static void
+gst_oss4_mixer_watch_process_changes (GstOss4Mixer * mixer)
+{
+ GList *c, *t, *tracks = NULL;
+
+ GST_INFO_OBJECT (mixer, "mixer interface or control changed");
+
+ /* this is all with the mixer object lock held */
+
+ /* we go through the list backwards so we can bail out faster when the entire
+ * interface needs to be rebuilt */
+ for (c = g_list_last (mixer->controls); c != NULL; c = c->prev) {
+ GstOss4MixerControl *mc = c->data;
+ oss_mixer_value ossval = { 0, };
+
+ mc->changed = FALSE;
+ mc->list_changed = FALSE;
+
+ /* not interested in controls we don't expose in the mixer interface */
+ if (!mc->used)
+ continue;
+
+ /* don't try to read a value from controls that don't have one */
+ if (mc->mixext.type == MIXT_DEVROOT || mc->mixext.type == MIXT_GROUP)
+ continue;
+
+ /* is this an enum control whose list may change? */
+ if (mc->mixext.type == MIXT_ENUM && mc->enum_version != 0) {
+ if (gst_oss4_mixer_enum_control_update_enum_list (mixer, mc))
+ mc->list_changed = TRUE;
+ }
+
+ ossval.dev = mc->mixext.dev;
+ ossval.ctrl = mc->mixext.ctrl;
+ ossval.timestamp = mc->mixext.timestamp;
+
+ if (ioctl (mixer->fd, SNDCTL_MIX_READ, &ossval) == -1) {
+ if (errno == EIDRM || errno == EFAULT) {
+ GST_DEBUG ("%s has disappeared", mc->mixext.extname);
+ goto mixer_changed;
+ }
+ GST_WARNING_OBJECT (mixer, "MIX_READ failed: %s", g_strerror (errno));
+ /* just ignore, move on to next one */
+ continue;
+ }
+
+ if (ossval.value == mc->last_val) { /* no change */
+ /* GST_LOG_OBJECT (mixer, "%s hasn't changed", mc->mixext.extname); */
+ continue;
+ }
+
+ mc->last_val = ossval.value;
+ GST_LOG_OBJECT (mixer, "%s changed value to %u 0x%08x",
+ mc->mixext.extname, ossval.value, ossval.value);
+ mc->changed = TRUE;
+ }
+
+ /* copy list and take track refs, so we can safely drop the object lock,
+ * which we need to do to be able to post messages on the bus */
+ tracks = g_list_copy (mixer->tracks);
+ g_list_foreach (tracks, (GFunc) g_object_ref, NULL);
+
+ GST_OBJECT_UNLOCK (mixer);
+
+ /* since we don't know (or want to know exactly) which controls belong to
+ * which track, we just go through the tracks one-by-one now and make them
+ * check themselves if any of their controls have changed and which messages
+ * to post on the bus as a result */
+ for (t = tracks; t != NULL; t = t->next) {
+ GstMixerTrack *track = t->data;
+
+ if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+ gst_oss4_mixer_slider_process_change_unlocked (track);
+ } else if (GST_IS_OSS4_MIXER_SWITCH (track)) {
+ gst_oss4_mixer_switch_process_change_unlocked (track);
+ } else if (GST_IS_OSS4_MIXER_ENUM (track)) {
+ gst_oss4_mixer_enum_process_change_unlocked (track);
+ }
+
+ g_object_unref (track);
+ }
+ g_list_free (tracks);
+
+ GST_OBJECT_LOCK (mixer);
+ return;
+
+mixer_changed:
+ {
+ GST_OBJECT_UNLOCK (mixer);
+ gst_mixer_mixer_changed (GST_MIXER (mixer));
+ GST_OBJECT_LOCK (mixer);
+ return;
+ }
+}
+
+/* This thread watches the mixer for changes in a somewhat inefficient way
+ * (running an ioctl every half second or so). This is still better and
+ * cheaper than apps polling all tracks for changes a few times a second
+ * though. Needs more thought. There's probably (hopefully) a way to get
+ * notifications via the fd directly somehow. */
+static gpointer
+gst_oss4_mixer_watch_thread (gpointer thread_data)
+{
+ GstOss4Mixer *mixer = GST_OSS4_MIXER_CAST (thread_data);
+
+ GST_DEBUG_OBJECT (mixer, "watch thread running");
+
+ GST_OBJECT_LOCK (mixer);
+ while (!mixer->watch_shutdown) {
+ oss_mixerinfo mi = { 0, };
+ GTimeVal tv;
+
+ mi.dev = -1;
+ if (ioctl (mixer->fd, SNDCTL_MIXERINFO, &mi) == 0) {
+ if (mixer->modify_counter != mi.modify_counter) {
+ /* GST_LOG ("processing changes"); */
+ gst_oss4_mixer_watch_process_changes (mixer);
+ mixer->modify_counter = mi.modify_counter;
+ } else {
+ /* GST_LOG ("no changes"); */
+ }
+ } else {
+ GST_WARNING_OBJECT (mixer, "MIXERINFO failed: %s", g_strerror (errno));
+ }
+
+ /* we could move the _get_current_time out of the loop and just do the
+ * add in ever iteration, which would be less exact, but who cares */
+ g_get_current_time (&tv);
+ g_time_val_add (&tv, GST_OSS4_MIXER_WATCH_INTERVAL * 1000);
+ g_cond_timed_wait (mixer->watch_cond, GST_OBJECT_GET_LOCK (mixer), &tv);
+ }
+ GST_OBJECT_UNLOCK (mixer);
+
+ GST_DEBUG_OBJECT (mixer, "watch thread done");
+ gst_object_unref (mixer);
+ return NULL;
+}
+
+/* call with object lock held */
+static void
+gst_oss4_mixer_wake_up_watch_task (GstOss4Mixer * mixer)
+{
+ GST_LOG_OBJECT (mixer, "signalling watch thread to wake up");
+ g_cond_signal (mixer->watch_cond);
+}
+
+static void
+gst_oss4_mixer_stop_watch_task (GstOss4Mixer * mixer)
+{
+ if (mixer->watch_thread) {
+ GST_OBJECT_LOCK (mixer);
+ mixer->watch_shutdown = TRUE;
+ GST_LOG_OBJECT (mixer, "signalling watch thread to stop");
+ g_cond_signal (mixer->watch_cond);
+ GST_OBJECT_UNLOCK (mixer);
+ GST_LOG_OBJECT (mixer, "waiting for watch thread to join");
+ g_thread_join (mixer->watch_thread);
+ GST_DEBUG_OBJECT (mixer, "watch thread stopped");
+ mixer->watch_thread = NULL;
+ }
+
+ if (mixer->watch_cond) {
+ g_cond_free (mixer->watch_cond);
+ mixer->watch_cond = NULL;
+ }
+}
+
+static void
+gst_oss4_mixer_start_watch_task (GstOss4Mixer * mixer)
+{
+ GError *err = NULL;
+
+ mixer->watch_cond = g_cond_new ();
+ mixer->watch_shutdown = FALSE;
+
+ mixer->watch_thread = g_thread_create (gst_oss4_mixer_watch_thread,
+ gst_object_ref (mixer), TRUE, &err);
+
+ if (mixer->watch_thread == NULL) {
+ GST_ERROR_OBJECT (mixer, "Could not create watch thread: %s", err->message);
+ g_cond_free (mixer->watch_cond);
+ mixer->watch_cond = NULL;
+ g_error_free (err);
+ }
+}
+
+static GstStateChangeReturn
+gst_oss4_mixer_change_state (GstElement * element, GstStateChange transition)
+{
+ GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+ GstOss4Mixer *mixer = GST_OSS4_MIXER (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ if (!gst_oss4_mixer_open (mixer, FALSE))
+ return GST_STATE_CHANGE_FAILURE;
+ gst_oss4_mixer_start_watch_task (mixer);
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ return ret;
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ gst_oss4_mixer_stop_watch_task (mixer);
+ gst_oss4_mixer_close (mixer);
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
+
+/* === GstMixer interface === */
+
+static inline gboolean
+gst_oss4_mixer_contains_track (GstMixer * mixer, GstMixerTrack * track)
+{
+ return (g_list_find (GST_OSS4_MIXER (mixer)->tracks, track) != NULL);
+}
+
+static inline gboolean
+gst_oss4_mixer_contains_options (GstMixer * mixer, GstMixerOptions * options)
+{
+ return (g_list_find (GST_OSS4_MIXER (mixer)->tracks, options) != NULL);
+}
+
+static void
+gst_oss4_mixer_post_mixer_changed_msg (GstOss4Mixer * mixer)
+{
+ /* only post mixer-changed message once */
+ if (!mixer->need_update) {
+ gst_mixer_mixer_changed (GST_MIXER (mixer));
+ mixer->need_update = TRUE;
+ }
+}
+
+/* call with mixer object lock held to serialise ioctl */
+gboolean
+gst_oss4_mixer_get_control_val (GstOss4Mixer * mixer, GstOss4MixerControl * mc,
+ int *val)
+{
+ oss_mixer_value ossval = { 0, };
+
+ if (GST_OBJECT_TRYLOCK (mixer)) {
+ GST_ERROR ("must be called with mixer lock held");
+ GST_OBJECT_UNLOCK (mixer);
+ }
+
+ ossval.dev = mc->mixext.dev;
+ ossval.ctrl = mc->mixext.ctrl;
+ ossval.timestamp = mc->mixext.timestamp;
+
+ if (ioctl (mixer->fd, SNDCTL_MIX_READ, &ossval) == -1) {
+ if (errno == EIDRM) {
+ GST_DEBUG_OBJECT (mixer, "MIX_READ failed: mixer interface has changed");
+ gst_oss4_mixer_post_mixer_changed_msg (mixer);
+ } else {
+ GST_WARNING_OBJECT (mixer, "MIX_READ failed: %s", g_strerror (errno));
+ }
+ *val = 0;
+ mc->last_val = 0;
+ return FALSE;
+ }
+
+ *val = ossval.value;
+ mc->last_val = ossval.value;
+ GST_LOG_OBJECT (mixer, "got value 0x%08x from %s)", *val, mc->mixext.extname);
+ return TRUE;
+}
+
+/* call with mixer object lock held to serialise ioctl */
+gboolean
+gst_oss4_mixer_set_control_val (GstOss4Mixer * mixer, GstOss4MixerControl * mc,
+ int val)
+{
+ oss_mixer_value ossval = { 0, };
+
+ ossval.dev = mc->mixext.dev;
+ ossval.ctrl = mc->mixext.ctrl;
+ ossval.timestamp = mc->mixext.timestamp;
+ ossval.value = val;
+
+ if (GST_OBJECT_TRYLOCK (mixer)) {
+ GST_ERROR ("must be called with mixer lock held");
+ GST_OBJECT_UNLOCK (mixer);
+ }
+
+ if (ioctl (mixer->fd, SNDCTL_MIX_WRITE, &ossval) == -1) {
+ if (errno == EIDRM) {
+ GST_LOG_OBJECT (mixer, "MIX_WRITE failed: mixer interface has changed");
+ gst_oss4_mixer_post_mixer_changed_msg (mixer);
+ } else {
+ GST_WARNING_OBJECT (mixer, "MIX_WRITE failed: %s", g_strerror (errno));
+ }
+ return FALSE;
+ }
+
+ mc->last_val = val;
+ GST_LOG_OBJECT (mixer, "set value 0x%08x on %s", val, mc->mixext.extname);
+ return TRUE;
+}
+
+static gchar *
+gst_oss4_mixer_control_get_pretty_name (GstOss4MixerControl * mc)
+{
+ gchar *name;
+
+#if 0
+ const gchar *name, *u;
+
+ /* "The id field is the original name given by the driver when it called
+ * mixer_ext_create_control. This name can be used by fully featured GUI
+ * mixers. However this name should be downshifted and cut before the last
+ * underscore ("_") to get the proper name. For example mixer control name
+ * created as "MYDRV_MAINVOL" will become just "mainvol" after this
+ * transformation." */
+ name = mc->mixext.extname;
+ u = MAX (strrchr (name, '_'), strrchr (name, '.'));
+ if (u != NULL)
+ name = u + 1;
+
+ /* maybe capitalize the first letter? */
+ return g_ascii_strdown (name, -1);
+#endif
+ /* the .id thing doesn't really seem to work right, ie. for some sliders
+ * it's just '-' so you have to use the name of the parent control etc.
+ * let's not use it for now, much too painful. */
+ if (g_str_has_prefix (mc->mixext.extname, "misc."))
+ name = g_strdup (mc->mixext.extname + 5);
+ else
+ name = g_strdup (mc->mixext.extname);
+ /* chop off trailing digit (only one for now) */
+ if (strlen (name) > 0 && g_ascii_isdigit (name[strlen (name) - 1]))
+ name[strlen (name) - 1] = '\0';
+ g_strdelimit (name, ".", ' ');
+ return name;
+}
+
+#if 0
+/* FIXME: translations for common option strings */
+static struct
+{
+ const gchar oss_name[32];
+ const gchar *label;
+} option_labels[] = {
+ {
+ "Fast", N_("Fast")}, {
+ "Low", N_("Low")}, {
+ "Medium", N_("Medium")}, {
+ "High", N_("High")}, {
+ "High+", N_("Very high")}, {
+ "Production", N_("Production")}, {
+ "OFF", N_("Off")}, {
+ "ON", N_("On")}, {
+ "Stereo", N_("Stereo")}, {
+ "Multich", N_("Surround sound")}, {
+ "input-mix", N_("Input mix")}, {
+ "front", N_("Front")}, {
+ "rear", N_("Rear")}, {
+ "side", N_("Side")}, {
+ "center/LFE", N_("Center / LFE")}, {
+ "mic", N_("Microphone")}, {
+ "fp-mic", N_("Front panel microphone")}, {
+ "input", N_("Input")}, {
+ "linein", N_("Line-in")}, {
+ "pcm1", N_("PCM 1")}, {
+ "pcm2", N_("PCM 2")}, {
+ "pcm3", N_("PCM 3")}, {
+"pcm4", N_("PCM 4")},};
+#endif
+
+/* these translations are a bit ad-hoc and horribly incomplete; it's not
+ * really going to work this way with all the different chipsets and drivers */
+static struct
+{
+ const gchar oss_name[32];
+ const gchar *label;
+} labels[] = {
+ /* connectors (e.g. hdaudio) */
+ {
+ "jack.green", N_("Green connector")}, {
+ "jack.fp-green", N_("Green front panel connector")}, {
+ "jack.pink", N_("Pink connector")}, {
+ "jack.fp-pink", N_("Pink front panel connector")}, {
+ "jack.blue", N_("Blue connector")}, {
+ "jack.fp-blue", N_("Blue front panel connector")}, {
+ "jack.orange", N_("Orange connector")}, {
+ "jack.fp-orange", N_("Orange front panel connector")}, {
+ "jack.black", N_("Black connector")}, {
+ "jack.fp-black", N_("Black front panel connector")}, {
+ "jack.gray", N_("Gray connector")}, {
+ "jack.fp-gray", N_("Gray front panel connector")}, {
+ "jack.white", N_("White connector")}, {
+ "jack.fp-white", N_("White front panel connector")}, {
+ "jack.red", N_("Red connector")}, {
+ "jack.fp-red", N_("Red front panel connector")}, {
+ "jack.yellow", N_("Yellow connector")}, {
+ "jack.fp-yellow", N_("Yellow front panel connector")},
+ /* connector functions (e.g. hdaudio) */
+ {
+ "jack.green.mode", N_("Green connector function")}, {
+ "jack.fp-green.mode", N_("Green front panel connector function")}, {
+ "jack.pink.mode", N_("Pink connector function")}, {
+ "jack.fp-pink.mode", N_("Pink front panel connector function")}, {
+ "jack.blue.mode", N_("Blue connector function")}, {
+ "jack.fp-blue.mode", N_("Blue front panel connector function")}, {
+ "jack.orange.mode", N_("Orange connector function")}, {
+ "jack.fp-orange.mode", N_("Orange front panel connector function")}, {
+ "jack.black.mode", N_("Black connector function")}, {
+ "jack.fp-black.mode", N_("Black front panel connector function")}, {
+ "jack.gray.mode", N_("Gray connector function")}, {
+ "jack.fp-gray.mode", N_("Gray front panel connector function")}, {
+ "jack.white.mode", N_("White connector function")}, {
+ "jack.fp-white.mode", N_("White front panel connector function")}, {
+ "jack.red.mode", N_("Red connector function")}, {
+ "jack.fp-red.mode", N_("Red front panel connector function")}, {
+ "jack.yellow.mode", N_("Yellow connector function")}, {
+ "jack.fp-yellow.mode", N_("Yellow front panel connector function")},
+ /* other */
+ {
+ "misc.mic", N_("Microphone")}, {
+ "misc.fp-mic", N_("Front panel microphone")}, {
+ "misc.linein", N_("Line-in")}, {
+ "misc.fp-linein", N_("Front panel line-in")}, {
+ "misc.headphone", N_("Headphones")}, {
+ "misc.fp-headphone", N_("Front panel headphones")}, {
+ "misc.front", N_("Front")}, {
+ "misc.rear", N_("Rear")}, {
+ "misc.side", N_("Side")}, {
+ "misc.center/lfe", N_("Center / LFE")}, {
+ "misc.pcm", N_("PCM")}, {
+ "misc.input-mix", N_("Input mix")}
+ /* FIXME translate Audigy NX USB labels) */
+/*
+ { "rec.src", N_("Record Source") },
+ { "output.mute", N_("Mute output") }
+ headph (Controller group)
+ headph.src (Enumeration control)
+ headph.mute (On/Off switch)
+ digital2 (Controller group)
+ digital2.src (Enumeration control)
+ digital2.mute (On/Off switch)
+ digital (Controller group)
+ digital.mute1 (On/Off switch)
+ digital.vol (Controller group)
+ digital.vol.front (Stereo slider (0-255))
+ digital.vol.surr (Stereo slider (0-255))
+ digital.vol.c/l (Stereo slider (0-255))
+ digital.vol.center (Stereo slider (0-255))
+ digital.mute2 (On/Off switch)
+ digital.vol (Stereo slider (0-255))
+ line (Controller group)
+ line.mute (On/Off switch)
+ line.vol (Stereo slider (0-255))
+ play-altset (Enumeration control)
+ rec-altset (Enumeration control)
+*/
+};
+
+/* Decent i18n is pretty much impossible with OSS's way of providing us with
+ * mixer labels (and the fact that they are pretty much random), but that
+ * doesn't mean we shouldn't at least try. */
+const gchar *
+gst_oss4_mixer_control_get_translated_name (GstOss4MixerControl * mc)
+{
+ gchar name[33] = { 0, };
+ char vmix_str[32] = { '\0', };
+ int dummy, i;
+
+ /* main virtual mixer controls (we hide the stream volumes) */
+ if (sscanf (mc->mixext.extname, "vmix%d-%32c", &dummy, vmix_str) == 2) {
+ if (strcmp (vmix_str, "src") == 0)
+ return _("Virtual mixer input");
+ else if (strcmp (vmix_str, "vol") == 0)
+ return _("Virtual mixer output");
+ else if (strcmp (vmix_str, "channels") == 0)
+ return _("Virtual mixer channel configuration");
+ }
+
+ /* munge connector.foo => jack.foo (change from 4.0 -> 4.1) */
+ if (g_str_has_prefix (mc->mixext.extname, "connector.")) {
+ g_snprintf (name, sizeof (name), "jack.%s", mc->mixext.extname + 10);
+ } else {
+ g_strlcpy (name, mc->mixext.extname, sizeof (name));
+ }
+
+ /* munge foo.function => foo.mode (change from 4.0 -> 4.1) */
+ if (g_str_has_suffix (name, ".function"))
+ memcpy (name + strlen (name) - strlen (".function"), ".mode", 5 + 1);
+
+ /* chop off trailing numbers */
+ while (strlen (name) > 0 && g_ascii_isdigit (name[strlen (name) - 1]))
+ name[strlen (name) - 1] = '\0';
+
+ for (i = 0; i < G_N_ELEMENTS (labels); ++i) {
+ if (strcmp (name, labels[i].oss_name) == 0)
+ return _(labels[i].label);
+ }
+
+ g_strdelimit (name, ".", ' ');
+ return g_quark_to_string (g_quark_from_string (name)); /* eek */
+}
+
+#ifndef GST_DISABLE_DEBUG
+static const gchar *
+mixer_ext_type_get_name (gint type)
+{
+ switch (type) {
+ case MIXT_DEVROOT:
+ return "Device root entry";
+ case MIXT_GROUP:
+ return "Controller group";
+ case MIXT_ONOFF:
+ return "On/Off switch";
+ case MIXT_ENUM:
+ return "Enumeration control";
+ case MIXT_MONOSLIDER:
+ return "Mono slider (0-255)";
+ case MIXT_STEREOSLIDER:
+ return "Stereo slider (0-255)";
+ case MIXT_MESSAGE:
+ return "Textual message"; /* whatever this is */
+ case MIXT_MONOVU:
+ return "Mono VU meter value";
+ case MIXT_STEREOVU:
+ return "Stereo VU meter value";
+ case MIXT_MONOPEAK:
+ return "Mono VU meter peak value";
+ case MIXT_STEREOPEAK:
+ return "Stereo VU meter peak value";
+ case MIXT_RADIOGROUP:
+ return "Radio button group";
+ case MIXT_MARKER: /* Separator between normal and extension entries */
+ return "Separator";
+ case MIXT_VALUE:
+ return "Decimal value entry";
+ case MIXT_HEXVALUE:
+ return "Hex value entry";
+ case MIXT_SLIDER:
+ return "Mono slider (31-bit value range)";
+ case MIXT_3D:
+ return "3D"; /* what's this? */
+ case MIXT_MONOSLIDER16:
+ return "Mono slider (0-32767)";
+ case MIXT_STEREOSLIDER16:
+ return "Stereo slider (0-32767)";
+ case MIXT_MUTE:
+ return "Mute switch";
+ default:
+ break;
+ }
+ return "unknown";
+}
+#endif /* GST_DISABLE_DEBUG */
+
+#ifndef GST_DISABLE_DEBUG
+static const gchar *
+mixer_ext_flags_get_string (gint flags)
+{
+ struct
+ {
+ gint flag;
+ gchar nick[16];
+ } all_flags[] = {
+ /* first the important ones */
+ {
+ MIXF_MAINVOL, "MAINVOL"}, {
+ MIXF_PCMVOL, "PCMVOL"}, {
+ MIXF_RECVOL, "RECVOL"}, {
+ MIXF_MONVOL, "MONVOL"}, {
+ MIXF_DESCR, "DESCR"},
+ /* now the rest in the right order */
+ {
+ MIXF_READABLE, "READABLE"}, {
+ MIXF_WRITEABLE, "WRITABLE"}, {
+ MIXF_POLL, "POLL"}, {
+ MIXF_HZ, "HZ"}, {
+ MIXF_STRING, "STRING"}, {
+ MIXF_DYNAMIC, "DYNAMIC"}, {
+ MIXF_OKFAIL, "OKFAIL"}, {
+ MIXF_FLAT, "FLAT"}, {
+ MIXF_LEGACY, "LEGACY"}, {
+ MIXF_CENTIBEL, "CENTIBEL"}, {
+ MIXF_DECIBEL, "DECIBEL"}, {
+ MIXF_WIDE, "WIDE"}
+ };
+ GString *s;
+ GQuark q;
+ gint i;
+
+ if (flags == 0)
+ return "None";
+
+ s = g_string_new ("");
+ for (i = 0; i < G_N_ELEMENTS (all_flags); ++i) {
+ if ((flags & all_flags[i].flag)) {
+ if (s->len > 0)
+ g_string_append (s, " | ");
+ g_string_append (s, all_flags[i].nick);
+ flags &= ~all_flags[i].flag; /* unset */
+ }
+ }
+
+ /* unknown flags? */
+ if (flags != 0) {
+ if (s->len > 0)
+ g_string_append (s, " | ");
+ g_string_append (s, "???");
+ }
+
+ /* we'll just put it into the global quark table (cheeky, eh?) */
+ q = g_quark_from_string (s->str);
+ g_string_free (s, TRUE);
+
+ return g_quark_to_string (q);
+}
+#endif /* GST_DISABLE_DEBUG */
+
+#ifndef GST_DISABLE_DEBUG
+static void
+gst_oss4_mixer_control_dump_tree (GstOss4MixerControl * mc, gint depth)
+{
+ GList *c;
+ gchar spaces[64];
+ gint i;
+
+ depth = MIN (sizeof (spaces) - 1, depth);
+ for (i = 0; i < depth; ++i)
+ spaces[i] = ' ';
+ spaces[i] = '\0';
+
+ GST_LOG ("%s%s (%s)", spaces, mc->mixext.extname,
+ mixer_ext_type_get_name (mc->mixext.type));
+ for (c = mc->children; c != NULL; c = c->next) {
+ GstOss4MixerControl *child_mc = (GstOss4MixerControl *) c->data;
+
+ gst_oss4_mixer_control_dump_tree (child_mc, depth + 2);
+ }
+}
+#endif /* GST_DISABLE_DEBUG */
+
+static GList *
+gst_oss4_mixer_get_controls (GstOss4Mixer * mixer)
+{
+ GstOss4MixerControl *root_mc = NULL;
+ oss_mixerinfo mi = { 0, };
+ GList *controls = NULL;
+ GList *l;
+ guint i;
+
+ /* Get info for currently open fd */
+ mi.dev = -1;
+ if (ioctl (mixer->fd, SNDCTL_MIXERINFO, &mi) == -1)
+ goto no_mixerinfo;
+
+ if (mi.nrext <= 0) {
+ GST_DEBUG ("mixer %s has no controls", mi.id);
+ return NULL;
+ }
+
+ GST_INFO ("Reading controls for mixer %s", mi.id);
+
+ for (i = 0; i < mi.nrext; ++i) {
+ GstOss4MixerControl *mc;
+ oss_mixext mix_ext = { 0, };
+
+ mix_ext.dev = mi.dev;
+ mix_ext.ctrl = i;
+
+ if (ioctl (mixer->fd, SNDCTL_MIX_EXTINFO, &mix_ext) == -1) {
+ GST_DEBUG ("SNDCTL_MIX_EXTINFO failed on mixer %s, control %d: %s",
+ mi.id, i, g_strerror (errno));
+ continue;
+ }
+
+ /* if this is the last one, save it for is-interface-up-to-date checking */
+ if (i == mi.nrext)
+ mixer->last_mixext = mix_ext;
+
+ mc = g_new0 (GstOss4MixerControl, 1);
+ mc->mixext = mix_ext;
+
+ /* both control_no and desc fields are pretty useless, ie. not always set,
+ * if ever, so not listed here */
+ GST_INFO ("Control %d", mix_ext.ctrl);
+ GST_INFO (" name : %s", mix_ext.extname);
+ GST_INFO (" type : %s (%d)", mixer_ext_type_get_name (mix_ext.type),
+ mix_ext.type);
+ GST_INFO (" flags : %s (0x%04x)",
+ mixer_ext_flags_get_string (mix_ext.flags), mix_ext.flags);
+ GST_INFO (" parent : %d", mix_ext.parent);
+
+ /* get tooltip (just for informational purposes for now) */
+ if (MIXEXT_HAS_DESCRIPTION (mix_ext)) {
+ oss_mixer_enuminfo desc = { 0, };
+
+ desc.dev = mix_ext.dev;
+ desc.ctrl = mix_ext.ctrl;
+ if (ioctl (mixer->fd, SNDCTL_MIX_DESCRIPTION, &desc) >= 0) {
+ /* "The string may contain multiple lines. The first line is the
+ * 'tooltip'. Optional subsequent lines may contain more detailed
+ * help text. Lines are separated by a linefeed character." */
+ g_strdelimit (&desc.strings[desc.strindex[0]], "\n\r", '\0');
+ GST_INFO (" tooltip: %s", &desc.strings[desc.strindex[0]]);
+ }
+ }
+
+ if (!MIXEXT_IS_ROOT (mix_ext)) {
+ /* find parent (we assume it comes in the list before the child) */
+ for (l = controls; l != NULL; l = l->next) {
+ GstOss4MixerControl *parent_mc = (GstOss4MixerControl *) l->data;
+
+ if (parent_mc->mixext.ctrl == mix_ext.parent) {
+ mc->parent = parent_mc;
+ parent_mc->children = g_list_append (parent_mc->children, mc);
+ break;
+ }
+ }
+ if (mc->parent == NULL) {
+ GST_ERROR_OBJECT (mixer, "couldn't find parent for control %d", i);
+ g_free (mc);
+ continue;
+ }
+ } else if (root_mc == NULL) {
+ root_mc = mc;
+ } else {
+ GST_WARNING_OBJECT (mixer, "two root controls?!");
+ }
+
+ controls = g_list_prepend (controls, mc);
+ }
+
+#ifndef GST_DISABLE_DEBUG
+ gst_oss4_mixer_control_dump_tree (root_mc, 0);
+#endif
+
+ return g_list_reverse (controls);
+
+/* ERRORS */
+no_mixerinfo:
+ {
+ GST_WARNING ("SNDCTL_MIXERINFO failed on mixer device %s: %s", mi.id,
+ g_strerror (errno));
+ return NULL;
+ }
+}
+
+static void
+gst_oss4_mixer_controls_guess_master (GstOss4Mixer * mixer,
+ const GList * controls)
+{
+ GstOss4MixerControl *firstpcm_mc = NULL;
+ GstOss4MixerControl *master_mc = NULL;
+ const GList *l;
+
+ for (l = controls; l != NULL; l = l->next) {
+ GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+ if (((mc->mixext.flags & MIXF_MAINVOL)) && master_mc == NULL) {
+ GST_INFO_OBJECT (mixer, "Master control: %s", mc->mixext.extname);
+ master_mc = mc;
+ break;
+ }
+ /* do we need to check if it's a slider type here? */
+ if (((mc->mixext.flags & MIXF_PCMVOL)) && firstpcm_mc == NULL) {
+ GST_INFO_OBJECT (mixer, "First PCM control: %s", mc->mixext.extname);
+ firstpcm_mc = mc;
+ }
+ }
+
+ /* if no control with MIXF_MAINVOL found, use first one with PCMVOL */
+ if (master_mc == NULL && firstpcm_mc != NULL) {
+ GST_INFO_OBJECT (mixer, "Marking first PCM control as master: %s",
+ firstpcm_mc->mixext.extname);
+ master_mc = firstpcm_mc;
+ }
+
+ if (master_mc != NULL)
+ master_mc->is_master = TRUE;
+}
+
+/* type: -1 for all types, otherwise just return siblings with requested type */
+static GList *
+gst_oss4_mixer_control_get_siblings (GstOss4MixerControl * mc, gint type)
+{
+ GList *s, *siblings = NULL;
+
+ if (mc->parent == NULL)
+ return NULL;
+
+ for (s = mc->parent->children; s != NULL; s = s->next) {
+ GstOss4MixerControl *sibling = (GstOss4MixerControl *) s->data;
+
+ if (mc != sibling && (type < 0 || sibling->mixext.type == type))
+ siblings = g_list_append (siblings, sibling);
+ }
+
+ return siblings;
+}
+
+static void
+gst_oss4_mixer_controls_find_sliders (GstOss4Mixer * mixer,
+ const GList * controls)
+{
+ const GList *l;
+
+ for (l = controls; l != NULL; l = l->next) {
+ GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+ GList *s, *siblings;
+
+ if (!MIXEXT_IS_SLIDER (mc->mixext) || mc->parent == NULL || mc->used)
+ continue;
+
+ mc->is_slider = TRUE;
+ mc->used = TRUE;
+
+ siblings = gst_oss4_mixer_control_get_siblings (mc, -1);
+
+ /* Note: the names can be misleading and may not reflect the actual
+ * hierarchy of the controls, e.g. it's possible that a slider is called
+ * connector.green and the mute control then connector.green.mute, but
+ * both controls are in fact siblings and both children of the group
+ * 'green' instead of mute being a child of connector.green as the naming
+ * would seem to suggest */
+ GST_LOG ("Slider: %s, parent=%s, %d siblings", mc->mixext.extname,
+ mc->parent->mixext.extname, g_list_length (siblings));
+
+ for (s = siblings; s != NULL; s = s->next) {
+ GstOss4MixerControl *sibling = (GstOss4MixerControl *) s->data;
+
+ GST_LOG (" %s (%s)", sibling->mixext.extname,
+ mixer_ext_type_get_name (sibling->mixext.type));
+
+ if (sibling->mixext.type == MIXT_MUTE ||
+ g_str_has_suffix (sibling->mixext.extname, ".mute")) {
+ /* simple case: slider with single mute sibling. We assume the .mute
+ * suffix in the name won't change - can't really do much else anyway */
+ if (sibling->mixext.type == MIXT_ONOFF ||
+ sibling->mixext.type == MIXT_MUTE) {
+ GST_LOG (" mute control for %s is %s", mc->mixext.extname,
+ sibling->mixext.extname);
+ mc->mute = sibling;
+ sibling->used = TRUE;
+ }
+ /* a group of .mute controls. We assume they are all switches here */
+ if (sibling->mixext.type == MIXT_GROUP) {
+ GList *m;
+
+ for (m = sibling->children; m != NULL; m = m->next) {
+ GstOss4MixerControl *grouped_sibling = m->data;
+
+ if (grouped_sibling->mixext.type == MIXT_ONOFF ||
+ grouped_sibling->mixext.type == MIXT_MUTE) {
+ GST_LOG (" %s is grouped mute control for %s",
+ grouped_sibling->mixext.extname, mc->mixext.extname);
+ mc->mute_group = g_list_append (mc->mute_group, grouped_sibling);
+ }
+ }
+
+ GST_LOG (" %s has a group of %d mute controls",
+ mc->mixext.extname, g_list_length (mc->mute_group));
+
+ /* we don't mark the individual mute controls as used, only the
+ * group control, as we still want individual switches for the
+ * individual controls */
+ sibling->used = TRUE;
+ }
+ }
+ }
+ g_list_free (siblings);
+ }
+}
+
+/* should be called with the mixer object lock held because of the ioctl;
+ * returns TRUE if the list was read the first time or modified */
+static gboolean
+gst_oss4_mixer_enum_control_update_enum_list (GstOss4Mixer * mixer,
+ GstOss4MixerControl * mc)
+{
+ oss_mixer_enuminfo ei = { 0, };
+ guint num_existing = 0;
+ int i;
+
+ /* count and existing values */
+ while (mc->enum_vals != NULL && mc->enum_vals[num_existing] != 0)
+ ++num_existing;
+
+ ei.dev = mc->mixext.dev;
+ ei.ctrl = mc->mixext.ctrl;
+
+ /* if we have create a generic list with numeric IDs already and the
+ * number of values hasn't changed, then there's not much to do here */
+ if (mc->no_list && mc->enum_vals != NULL &&
+ num_existing == mc->mixext.maxvalue) {
+ return FALSE;
+ }
+
+ /* if we have a list and it doesn't change, there's nothing to do either */
+ if (mc->enum_vals != NULL && mc->enum_version == 0)
+ return FALSE;
+
+ if (ioctl (mixer->fd, SNDCTL_MIX_ENUMINFO, &ei) == -1) {
+ g_free (mc->enum_vals);
+ mc->enum_vals = g_new0 (GQuark, mc->mixext.maxvalue + 1);
+
+ GST_DEBUG ("no enum info available, creating numeric values from 0-%d",
+ mc->mixext.maxvalue - 1);
+
+ /* "It is possible that some enum controls don't have any name list
+ * available. In this case the application should automatically generate
+ * list of numbers (0 to N-1)" */
+ for (i = 0; i < mc->mixext.maxvalue; ++i) {
+ gchar num_str[8];
+
+ g_snprintf (num_str, sizeof (num_str), "%d", i);
+ mc->enum_vals[i] = g_quark_from_string (num_str);
+ }
+ mc->enum_version = 0; /* the only way to change is via maxvalue */
+ } else {
+ /* old list same as current list? */
+ if (mc->enum_vals != NULL && mc->enum_version == ei.version)
+ return FALSE;
+
+ /* no list yet, or the list has changed */
+ GST_LOG ("%s", (mc->enum_vals) ? "enum list has changed" : "reading list");
+ if (ei.nvalues != mc->mixext.maxvalue) {
+ GST_WARNING_OBJECT (mixer, "Enum: %s, nvalues %d != maxvalue %d",
+ mc->mixext.extname, ei.nvalues, mc->mixext.maxvalue);
+ mc->mixext.maxvalue = MIN (ei.nvalues, mc->mixext.maxvalue);
+ }
+
+ mc->mixext.maxvalue = MIN (mc->mixext.maxvalue, OSS_ENUM_MAXVALUE);
+
+ g_free (mc->enum_vals);
+ mc->enum_vals = g_new0 (GQuark, mc->mixext.maxvalue + 1);
+ for (i = 0; i < mc->mixext.maxvalue; ++i) {
+ GST_LOG (" %s", ei.strings + ei.strindex[i]);
+ mc->enum_vals[i] = g_quark_from_string (ei.strings + ei.strindex[i]);
+ }
+ }
+
+ return TRUE;
+}
+
+static void
+gst_oss4_mixer_controls_find_enums (GstOss4Mixer * mixer,
+ const GList * controls)
+{
+ const GList *l;
+
+ for (l = controls; l != NULL; l = l->next) {
+ GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+ if (mc->mixext.type != MIXT_ENUM || mc->used)
+ continue;
+
+ mc->is_enum = TRUE;
+ mc->used = TRUE;
+
+ /* Note: enums are special: for most controls, the maxvalue is inclusive,
+ * but for enum controls it's actually exclusive (boggle), so that
+ * mixext.maxvalue = num_values */
+
+ GST_LOG ("Enum: %s, parent=%s, num_enums=%d", mc->mixext.extname,
+ mc->parent->mixext.extname, mc->mixext.maxvalue);
+
+ gst_oss4_mixer_enum_control_update_enum_list (mixer, mc);
+ }
+}
+
+static void
+gst_oss4_mixer_controls_find_switches (GstOss4Mixer * mixer,
+ const GList * controls)
+{
+ const GList *l;
+
+ for (l = controls; l != NULL; l = l->next) {
+ GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+ if (mc->used)
+ continue;
+
+ if (mc->mixext.type != MIXT_ONOFF && mc->mixext.type != MIXT_MUTE)
+ continue;
+
+ mc->is_switch = TRUE;
+ mc->used = TRUE;
+
+ GST_LOG ("Switch: %s, parent=%s", mc->mixext.extname,
+ mc->parent->mixext.extname);
+ }
+}
+
+static void
+gst_oss4_mixer_controls_find_virtual (GstOss4Mixer * mixer,
+ const GList * controls)
+{
+ const GList *l;
+
+ for (l = controls; l != NULL; l = l->next) {
+ GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+ /* or sscanf (mc->mixext.extname, "vmix%d-out.", &n) == 1 ? */
+ /* for now we just flag all virtual controls with managed labels, those
+ * are really more appropriate for a pavucontrol-type control thing than
+ * the (more hardware-oriented) mixer interface */
+ if (mc->mixext.id[0] == '@') {
+ mc->is_virtual = TRUE;
+ GST_LOG ("%s is virtual control with managed label", mc->mixext.extname);
+ }
+ }
+}
+
+static void
+gst_oss4_mixer_controls_dump_unused (GstOss4Mixer * mixer,
+ const GList * controls)
+{
+ const GList *l;
+
+ for (l = controls; l != NULL; l = l->next) {
+ GstOss4MixerControl *mc = (GstOss4MixerControl *) l->data;
+
+ if (mc->used)
+ continue;
+
+ switch (mc->mixext.type) {
+ case MIXT_DEVROOT:
+ case MIXT_GROUP:
+ case MIXT_MESSAGE:
+ case MIXT_MONOVU:
+ case MIXT_STEREOVU:
+ case MIXT_MONOPEAK:
+ case MIXT_STEREOPEAK:
+ case MIXT_MARKER:
+ continue; /* not interested in these types of controls */
+ case MIXT_MONODB:
+ case MIXT_STEREODB:
+ GST_DEBUG ("obsolete control type %d", mc->mixext.type);
+ continue;
+ case MIXT_MONOSLIDER:
+ case MIXT_STEREOSLIDER:
+ case MIXT_SLIDER:
+ case MIXT_MONOSLIDER16:
+ case MIXT_STEREOSLIDER16:
+ /* this shouldn't happen */
+ GST_ERROR ("unused slider control?!");
+ continue;
+ case MIXT_VALUE:
+ case MIXT_HEXVALUE:
+ /* value entry, not sure what to do with that, skip for now */
+ continue;
+ case MIXT_ONOFF:
+ case MIXT_MUTE:
+ case MIXT_ENUM:
+ case MIXT_3D:
+ case MIXT_RADIOGROUP:
+ GST_DEBUG ("FIXME: handle %s %s",
+ mixer_ext_type_get_name (mc->mixext.type), mc->mixext.extname);
+ break;
+ default:
+ GST_WARNING ("unknown control type %d", mc->mixext.type);
+ continue;
+ }
+ }
+}
+
+static GList *
+gst_oss4_mixer_create_tracks (GstOss4Mixer * mixer, const GList * controls)
+{
+ const GList *c;
+ GList *tracks = NULL;
+
+ for (c = controls; c != NULL; c = c->next) {
+ GstOss4MixerControl *mc = (GstOss4MixerControl *) c->data;
+ GstMixerTrack *track = NULL;
+
+ if (mc->is_virtual)
+ continue;
+
+ if (mc->is_slider) {
+ track = gst_oss4_mixer_slider_new (mixer, mc);
+ } else if (mc->is_enum) {
+ track = gst_oss4_mixer_enum_new (mixer, mc);
+ } else if (mc->is_switch) {
+ track = gst_oss4_mixer_switch_new (mixer, mc);
+ }
+
+ if (track == NULL)
+ continue;
+
+ track->label = gst_oss4_mixer_control_get_pretty_name (mc);
+ track->flags = 0;
+
+ GST_LOG ("translated label: %s [%s] = %s", track->label, mc->mixext.id,
+ gst_oss4_mixer_control_get_translated_name (mc));
+
+ /* This whole 'a track is either INPUT or OUTPUT' model is just flawed,
+ * esp. if a slider's role can be changed on the fly, like when you change
+ * function of a connector. What should we do in that case? Change the flag
+ * and make the app rebuild the interface? Ignore it? */
+ if (g_str_has_prefix (mc->mixext.extname, "record.")) {
+ mc->is_output = FALSE;
+ mc->is_input = TRUE;
+ }
+
+ /* FIXME: determine is_input and is_output */
+ /* must be either INPUT or OUTPUT (but not both and not neither) for now,
+ * or gnome-volume-control aborts */
+ if (mc->is_input)
+ track->flags |= GST_MIXER_TRACK_INPUT;
+ else if (mc->is_output)
+ track->flags |= GST_MIXER_TRACK_OUTPUT;
+ else
+ track->flags |= GST_MIXER_TRACK_OUTPUT;
+
+ if (mc->is_master)
+ track->flags |= GST_MIXER_TRACK_MASTER;
+
+ tracks = g_list_append (tracks, track);
+ }
+
+ return tracks;
+}
+
+static void
+gst_oss4_mixer_update_tracks (GstOss4Mixer * mixer)
+{
+ GList *controls, *tracks;
+
+ /* read and process controls */
+ controls = gst_oss4_mixer_get_controls (mixer);
+
+ gst_oss4_mixer_controls_guess_master (mixer, controls);
+
+ gst_oss4_mixer_controls_find_sliders (mixer, controls);
+
+ gst_oss4_mixer_controls_find_enums (mixer, controls);
+
+ gst_oss4_mixer_controls_find_switches (mixer, controls);
+
+ gst_oss4_mixer_controls_find_virtual (mixer, controls);
+
+ gst_oss4_mixer_controls_dump_unused (mixer, controls);
+
+ tracks = gst_oss4_mixer_create_tracks (mixer, controls);
+
+ /* free old tracks and controls */
+ gst_oss4_mixer_free_tracks (mixer);
+
+ /* replace old with new */
+ mixer->tracks = tracks;
+ mixer->controls = controls;
+}
+
+static const GList *
+gst_oss4_mixer_list_tracks (GstMixer * mixer_iface)
+{
+ GstOss4Mixer *mixer = GST_OSS4_MIXER (mixer_iface);
+
+ g_return_val_if_fail (mixer != NULL, NULL);
+ g_return_val_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer), NULL);
+
+ GST_OBJECT_LOCK (mixer);
+
+ /* Do a read on the last control to check if the interface has changed */
+ if (!mixer->need_update && mixer->last_mixext.ctrl > 0) {
+ GstOss4MixerControl mc = { {0,}
+ ,
+ };
+ int val;
+
+ mc.mixext = mixer->last_mixext;
+ gst_oss4_mixer_get_control_val (mixer, &mc, &val);
+ }
+
+ if (mixer->need_update || mixer->tracks == NULL) {
+ gst_oss4_mixer_update_tracks (mixer);
+ mixer->need_update = FALSE;
+ }
+
+ GST_OBJECT_UNLOCK (mixer);
+
+ return (const GList *) mixer->tracks;
+}
+
+static void
+gst_oss4_mixer_set_volume (GstMixer * mixer, GstMixerTrack * track,
+ gint * volumes)
+{
+ GstOss4Mixer *oss;
+
+ g_return_if_fail (mixer != NULL);
+ g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+ g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+ g_return_if_fail (gst_oss4_mixer_contains_track (mixer, track));
+ g_return_if_fail (volumes != NULL);
+
+ oss = GST_OSS4_MIXER (mixer);
+
+ GST_OBJECT_LOCK (oss);
+
+ if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+ gst_oss4_mixer_slider_set_volume (GST_OSS4_MIXER_SLIDER (track), volumes);
+ }
+
+ GST_OBJECT_UNLOCK (oss);
+}
+
+static void
+gst_oss4_mixer_get_volume (GstMixer * mixer, GstMixerTrack * track,
+ gint * volumes)
+{
+ GstOss4Mixer *oss;
+
+ g_return_if_fail (mixer != NULL);
+ g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+ g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+ g_return_if_fail (gst_oss4_mixer_contains_track (mixer, track));
+ g_return_if_fail (volumes != NULL);
+
+ oss = GST_OSS4_MIXER (mixer);
+
+ GST_OBJECT_LOCK (oss);
+
+ memset (volumes, 0, track->num_channels * sizeof (gint));
+
+ if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+ gst_oss4_mixer_slider_get_volume (GST_OSS4_MIXER_SLIDER (track), volumes);
+ }
+
+ GST_OBJECT_UNLOCK (oss);
+}
+
+static void
+gst_oss4_mixer_set_record (GstMixer * mixer, GstMixerTrack * track,
+ gboolean record)
+{
+ GstOss4Mixer *oss;
+
+ g_return_if_fail (mixer != NULL);
+ g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+ g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+ g_return_if_fail (gst_oss4_mixer_contains_track (mixer, track));
+
+ oss = GST_OSS4_MIXER (mixer);
+
+ GST_OBJECT_LOCK (oss);
+
+ if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+ gst_oss4_mixer_slider_set_record (GST_OSS4_MIXER_SLIDER (track), record);
+ } else if (GST_IS_OSS4_MIXER_SWITCH (track)) {
+ if ((track->flags & GST_MIXER_TRACK_INPUT)) {
+ gst_oss4_mixer_switch_set (GST_OSS4_MIXER_SWITCH (track), record);
+ } else {
+ GST_WARNING_OBJECT (track, "set_record called on non-INPUT track");
+ }
+ }
+
+ GST_OBJECT_UNLOCK (oss);
+}
+
+static void
+gst_oss4_mixer_set_mute (GstMixer * mixer, GstMixerTrack * track, gboolean mute)
+{
+ GstOss4Mixer *oss;
+
+ g_return_if_fail (mixer != NULL);
+ g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+ g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+ g_return_if_fail (gst_oss4_mixer_contains_track (mixer, track));
+
+ oss = GST_OSS4_MIXER (mixer);
+
+ GST_OBJECT_LOCK (oss);
+
+ if (GST_IS_OSS4_MIXER_SLIDER (track)) {
+ gst_oss4_mixer_slider_set_mute (GST_OSS4_MIXER_SLIDER (track), mute);
+ } else if (GST_IS_OSS4_MIXER_SWITCH (track)) {
+ if ((track->flags & GST_MIXER_TRACK_OUTPUT)) {
+ gst_oss4_mixer_switch_set (GST_OSS4_MIXER_SWITCH (track), mute);
+ } else {
+ GST_WARNING_OBJECT (track, "set_mute called on non-OUTPUT track");
+ }
+ }
+
+ GST_OBJECT_UNLOCK (oss);
+}
+
+static void
+gst_oss4_mixer_set_option (GstMixer * mixer, GstMixerOptions * options,
+ gchar * value)
+{
+ GstOss4Mixer *oss;
+
+ g_return_if_fail (mixer != NULL);
+ g_return_if_fail (value != NULL);
+ g_return_if_fail (GST_IS_OSS4_MIXER (mixer));
+ g_return_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer));
+ g_return_if_fail (GST_IS_OSS4_MIXER_ENUM (options));
+ g_return_if_fail (gst_oss4_mixer_contains_options (mixer, options));
+
+ oss = GST_OSS4_MIXER (mixer);
+
+ GST_OBJECT_LOCK (oss);
+
+ if (!gst_oss4_mixer_enum_set_option (GST_OSS4_MIXER_ENUM (options), value)) {
+ /* not much we can do here but wake up the watch thread early, so it
+ * can do its thing and post messages if anything has changed */
+ gst_oss4_mixer_wake_up_watch_task (oss);
+ }
+
+ GST_OBJECT_UNLOCK (oss);
+}
+
+static const gchar *
+gst_oss4_mixer_get_option (GstMixer * mixer, GstMixerOptions * options)
+{
+ GstOss4Mixer *oss;
+ const gchar *current_val;
+
+ g_return_val_if_fail (mixer != NULL, NULL);
+ g_return_val_if_fail (GST_IS_OSS4_MIXER (mixer), NULL);
+ g_return_val_if_fail (GST_OSS4_MIXER_IS_OPEN (mixer), NULL);
+ g_return_val_if_fail (GST_IS_OSS4_MIXER_ENUM (options), NULL);
+ g_return_val_if_fail (gst_oss4_mixer_contains_options (mixer, options), NULL);
+
+ oss = GST_OSS4_MIXER (mixer);
+
+ GST_OBJECT_LOCK (oss);
+
+ current_val = gst_oss4_mixer_enum_get_option (GST_OSS4_MIXER_ENUM (options));
+
+ if (current_val == NULL) {
+ /* not much we can do here but wake up the watch thread early, so it
+ * can do its thing and post messages if anything has changed */
+ gst_oss4_mixer_wake_up_watch_task (oss);
+ }
+
+ GST_OBJECT_UNLOCK (oss);
+
+ return current_val;
+}
+
+static GstMixerFlags
+gst_oss4_mixer_get_mixer_flags (GstMixer * mixer)
+{
+ return GST_MIXER_FLAG_AUTO_NOTIFICATIONS;
+}
+
+static void
+gst_oss4_mixer_interface_init (GstMixerClass * klass)
+{
+ GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE;
+
+ klass->list_tracks = gst_oss4_mixer_list_tracks;
+ klass->set_volume = gst_oss4_mixer_set_volume;
+ klass->get_volume = gst_oss4_mixer_get_volume;
+ klass->set_mute = gst_oss4_mixer_set_mute;
+ klass->set_record = gst_oss4_mixer_set_record;
+ klass->set_option = gst_oss4_mixer_set_option;
+ klass->get_option = gst_oss4_mixer_get_option;
+ klass->get_mixer_flags = gst_oss4_mixer_get_mixer_flags;
+}
+
+/* Implement the horror that is GstImplementsInterface */
+
+static gboolean
+gst_oss4_mixer_supported (GstImplementsInterface * iface, GType iface_type)
+{
+ GstOss4Mixer *mixer;
+
+ g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
+
+ mixer = GST_OSS4_MIXER (iface);
+
+ return GST_OSS4_MIXER_IS_OPEN (mixer);
+}
+
+static void
+gst_oss4_mixer_implements_interface_init (GstImplementsInterfaceClass * klass)
+{
+ klass->supported = gst_oss4_mixer_supported;
+}
+
+static void
+gst_oss4_mixer_init_interfaces (GType type)
+{
+ static const GInterfaceInfo implements_iface_info = {
+ (GInterfaceInitFunc) gst_oss4_mixer_implements_interface_init,
+ NULL,
+ NULL,
+ };
+ static const GInterfaceInfo mixer_iface_info = {
+ (GInterfaceInitFunc) gst_oss4_mixer_interface_init,
+ NULL,
+ NULL,
+ };
+
+ g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
+ &implements_iface_info);
+ g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
+
+ gst_oss4_add_property_probe_interface (type);
+}
--- /dev/null
+/* GStreamer OSS4 mixer implementation
+ * 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.
+ */
+
+#ifndef OSS4_MIXER_H
+#define OSS4_MIXER_H
+
+#include <gst/gst.h>
+
+#include "oss4-soundcard.h"
+
+#define GST_OSS4_MIXER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_MIXER,GstOss4Mixer))
+#define GST_OSS4_MIXER_CAST(obj) ((GstOss4Mixer *)(obj))
+#define GST_OSS4_MIXER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_MIXER,GstOss4MixerClass))
+#define GST_IS_OSS4_MIXER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_MIXER))
+#define GST_IS_OSS4_MIXER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_MIXER))
+#define GST_TYPE_OSS4_MIXER (gst_oss4_mixer_get_type())
+
+#define GST_OSS4_MIXER_IS_OPEN(obj) (GST_OSS4_MIXER(obj)->fd != -1)
+
+typedef struct _GstOss4Mixer GstOss4Mixer;
+typedef struct _GstOss4MixerClass GstOss4MixerClass;
+
+struct _GstOss4Mixer {
+ GstElement element;
+
+ /*< private >*/
+
+ /* element bits'n'bops */
+ gchar * device;
+
+ /* mixer details */
+ gint fd; /* file descriptor if open, or -1 */
+ gchar * device_name; /* device description, or NULL */
+ gchar * open_device; /* the device we opened */
+
+ GList * tracks; /* list of available tracks */
+ GList * controls; /* list of available controls */
+ gboolean need_update; /* re-read list of available tracks? */
+
+ oss_mixext last_mixext; /* we keep this around so we can
+ * easily check if the mixer
+ * interface has changed */
+
+ GThread * watch_thread; /* thread watching for value changes */
+ GCond * watch_cond;
+ gint watch_shutdown;
+ gint modify_counter; /* from MIXERINFO */
+
+ /* for property probe interface */
+ GList * property_probe_list;
+};
+
+struct _GstOss4MixerClass {
+ GstElementClass element_class;
+};
+
+/* helper struct holding info about one control */
+typedef struct _GstOss4MixerControl GstOss4MixerControl;
+
+struct _GstOss4MixerControl {
+ oss_mixext mixext;
+ GstOss4MixerControl *parent; /* NULL if root */
+ GstOss4MixerControl *mute; /* sibling with mute function, or NULL */
+ GList *mute_group; /* group of mute controls, or NULL */
+ GList *children; /* GstOss4MixerControls (no ownership) */
+
+ GQuark *enum_vals; /* 0-terminated array of values or NULL */
+ int enum_version; /* 0 = list won't change */
+
+ int last_val; /* last value seen */
+
+ gboolean is_virtual : 1; /* is a vmix control with dynamic label */
+ gboolean is_master : 1;
+ gboolean is_slider : 1; /* represent as slider */
+ gboolean is_switch : 1; /* represent as switch */
+ gboolean is_enum : 1; /* represent as combo/enumeration */
+ gboolean no_list : 1; /* enumeration with no list available */
+ gboolean is_input : 1; /* is an input-related control */
+ gboolean is_output : 1; /* is an output-related control */
+ gboolean used : 1; /* whether we know what to do with this */
+
+ gboolean changed : 1; /* transient flag used by watch thread */
+ gboolean list_changed : 1; /* transient flag used by watch thread */
+};
+
+/* header says parent=-1 means root, but it can also be parent=ctrl */
+#define MIXEXT_IS_ROOT(me) ((me).parent == -1 || (me).parent == (me).ctrl)
+
+#define MIXEXT_IS_SLIDER(me) ((me).type == MIXT_MONOSLIDER || \
+ (me).type == MIXT_STEREOSLIDER || (me).type == MIXT_MONOSLIDER16 || \
+ (me).type == MIXT_STEREOSLIDER16 || (me).type == MIXT_SLIDER)
+
+#define MIXEXT_HAS_DESCRIPTION(me) (((me).flags & MIXF_DESCR) != 0)
+
+#define MIXEXT_ENUM_IS_AVAILABLE(me,num) \
+ (((me).enum_present[num/8]) & (1 << (num % 8)))
+
+
+GType gst_oss4_mixer_get_type (void);
+
+gboolean gst_oss4_mixer_get_control_val (GstOss4Mixer * mixer,
+ GstOss4MixerControl * mc,
+ int * val);
+
+gboolean gst_oss4_mixer_set_control_val (GstOss4Mixer * mixer,
+ GstOss4MixerControl * mc,
+ int val);
+
+G_END_DECLS
+
+#endif /* OSS4_MIXER_H */
+
--- /dev/null
+/* GStreamer OSS4 audio property probe interface implementation
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-audio.h"
+#include "oss4-mixer.h"
+#include "oss4-sink.h"
+#include "oss4-source.h"
+#include "oss4-soundcard.h"
+#include "oss4-property-probe.h"
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <errno.h>
+#include <string.h>
+
+GST_DEBUG_CATEGORY_EXTERN (oss4_debug);
+#define GST_CAT_DEFAULT oss4_debug
+
+static const GList *
+gst_oss4_property_probe_get_properties (GstPropertyProbe * probe)
+{
+ GObjectClass *klass = G_OBJECT_GET_CLASS (probe);
+ GList *list;
+
+ GST_OBJECT_LOCK (GST_OBJECT (probe));
+
+ /* we create a new list and store it in the instance struct, since apparently
+ * we forgot to update the API for 0.10 (and why don't we mark probable
+ * properties with a flag instead anyway?). A bit hackish, but what can you
+ * do (can't really use a static variable since the pspec will be different
+ * for src and sink class); this isn't particularly pretty, but the best
+ * we can do given that we can't create a common base class (we could do
+ * fancy things with the interface, or use g_object_set_data instead, but
+ * it's not really going to make it much better) */
+ if (GST_IS_AUDIO_SINK_CLASS (klass)) {
+ list = GST_OSS4_SINK (probe)->property_probe_list;
+ } else if (GST_IS_AUDIO_SRC_CLASS (klass)) {
+ list = GST_OSS4_SOURCE (probe)->property_probe_list;
+ } else if (GST_IS_OSS4_MIXER_CLASS (klass)) {
+ list = GST_OSS4_MIXER (probe)->property_probe_list;
+ } else {
+ GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+ g_return_val_if_reached (NULL);
+ }
+
+ if (list == NULL) {
+ GParamSpec *pspec;
+
+ pspec = g_object_class_find_property (klass, "device");
+ list = g_list_prepend (NULL, pspec);
+
+ if (GST_IS_AUDIO_SINK_CLASS (klass)) {
+ GST_OSS4_SINK (probe)->property_probe_list = list;
+ } else if (GST_IS_AUDIO_SRC_CLASS (klass)) {
+ GST_OSS4_SOURCE (probe)->property_probe_list = list;
+ } else if (GST_IS_OSS4_MIXER_CLASS (klass)) {
+ GST_OSS4_MIXER (probe)->property_probe_list = list;
+ }
+ }
+
+ GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+
+ return list;
+}
+
+static void
+gst_oss4_property_probe_probe_property (GstPropertyProbe * probe,
+ guint prop_id, const GParamSpec * pspec)
+{
+ if (!g_str_equal (pspec->name, "device")) {
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (probe, prop_id, pspec);
+ }
+}
+
+static gboolean
+gst_oss4_property_probe_needs_probe (GstPropertyProbe * probe,
+ guint prop_id, const GParamSpec * pspec)
+{
+ /* don't cache probed data */
+ return TRUE;
+}
+
+static gint
+oss4_mixerinfo_priority_cmp (struct oss_mixerinfo *mi1,
+ struct oss_mixerinfo *mi2)
+{
+ /* return negative vaue if mi1 comes before mi2 */
+ if (mi1->priority != mi2->priority)
+ return mi2->priority - mi1->priority;
+
+ return strcmp (mi1->devnode, mi2->devnode);
+}
+
+/* caller must ensure LOCK is taken (e.g. if ioctls need to be serialised) */
+gboolean
+gst_oss4_property_probe_find_device_name (GstObject * obj, int fd,
+ const gchar * device_handle, gchar ** device_name)
+{
+ struct oss_sysinfo si = { {0,}, };
+ gchar *name = NULL;
+
+ if (ioctl (fd, SNDCTL_SYSINFO, &si) == 0) {
+ int i;
+
+ for (i = 0; i < si.numaudios; ++i) {
+ struct oss_audioinfo ai = { 0, };
+
+ ai.dev = i;
+ if (ioctl (fd, SNDCTL_AUDIOINFO, &ai) == -1) {
+ GST_DEBUG_OBJECT (obj, "AUDIOINFO ioctl for device %d failed", i);
+ continue;
+ }
+ if (strcmp (ai.devnode, device_handle) == 0) {
+ name = g_strdup (ai.name);
+ break;
+ }
+ }
+ } else {
+ GST_WARNING_OBJECT (obj, "SYSINFO ioctl failed: %s", g_strerror (errno));
+ }
+
+ /* try ENGINEINFO as fallback (which is better than nothing) */
+ if (name == NULL) {
+ struct oss_audioinfo ai = { 0, };
+
+ GST_LOG_OBJECT (obj, "device %s not listed in AUDIOINFO", device_handle);
+ ai.dev = -1;
+ if (ioctl (fd, SNDCTL_ENGINEINFO, &ai) == 0)
+ name = g_strdup (ai.name);
+ }
+
+ GST_DEBUG_OBJECT (obj, "Device name: %s", GST_STR_NULL (name));
+
+ if (name != NULL)
+ *device_name = name;
+
+ return (name != NULL);
+}
+
+static GList *
+gst_oss4_property_probe_get_mixer_devices (GstObject * obj, int fd,
+ struct oss_sysinfo *si)
+{
+ GList *m, *mixers = NULL;
+ GList *devices = NULL;
+
+ int i;
+
+ GST_LOG_OBJECT (obj, "%d mixer devices", si->nummixers);
+
+ /* first, find suitable mixer devices and sort by priority */
+ for (i = 0; i < si->nummixers; ++i) {
+ struct oss_mixerinfo mi = { 0, };
+
+ mi.dev = i;
+ if (ioctl (fd, SNDCTL_MIXERINFO, &mi) == -1) {
+ GST_DEBUG_OBJECT (obj, "MIXERINFO ioctl for device %d failed", i);
+ continue;
+ }
+
+ GST_INFO_OBJECT (obj, "mixer device %d:", i);
+ GST_INFO_OBJECT (obj, " enabled : %s", (mi.enabled) ? "yes" :
+ "no (powered off or unplugged)");
+ GST_INFO_OBJECT (obj, " priority : %d", mi.priority);
+ GST_INFO_OBJECT (obj, " devnode : %s", mi.devnode);
+ GST_INFO_OBJECT (obj, " handle : %s", mi.handle);
+ GST_INFO_OBJECT (obj, " caps : 0x%02x", mi.caps);
+ GST_INFO_OBJECT (obj, " name : %s", mi.name);
+
+ if (!mi.enabled) {
+ GST_DEBUG_OBJECT (obj, "mixer device is not usable/enabled, skipping");
+ continue;
+ }
+
+ /* only want mixers that control hardware directly */
+ if ((mi.caps & MIXER_CAP_VIRTUAL)) {
+ GST_DEBUG_OBJECT (obj, "mixer device is a virtual device, skipping");
+ continue;
+ }
+
+ mixers = g_list_insert_sorted (mixers, g_memdup (&mi, sizeof (mi)),
+ (GCompareFunc) oss4_mixerinfo_priority_cmp);
+ }
+
+ /* then create device list according to priority */
+ for (m = mixers; m != NULL; m = m->next) {
+ struct oss_mixerinfo *mi = (struct oss_mixerinfo *) m->data;
+
+ GST_LOG_OBJECT (obj, "mixer device: '%s'", mi->devnode);
+ devices = g_list_prepend (devices, g_strdup (mi->devnode));
+ g_free (m->data);
+ }
+ g_list_free (mixers);
+ mixers = NULL;
+
+ return g_list_reverse (devices);
+}
+
+static GList *
+gst_oss4_property_probe_get_audio_devices (GstObject * obj, int fd,
+ struct oss_sysinfo *si, int cap_mask)
+{
+ GList *devices = NULL;
+ int i;
+
+ GST_LOG_OBJECT (obj, "%d audio/dsp devices", si->numaudios);
+
+ for (i = 0; i < si->numaudios; ++i) {
+ struct oss_audioinfo ai = { 0, };
+
+ ai.dev = i;
+ if (ioctl (fd, SNDCTL_AUDIOINFO, &ai) == -1) {
+ GST_DEBUG_OBJECT (obj, "AUDIOINFO ioctl for device %d failed", i);
+ continue;
+ }
+
+ if ((ai.caps & cap_mask) == 0) {
+ GST_DEBUG_OBJECT (obj, "audio device %d is not an %s device", i,
+ (cap_mask == PCM_CAP_OUTPUT) ? "output" : "input");
+ continue;
+ }
+
+ if (!ai.enabled) {
+ GST_DEBUG_OBJECT (obj, "audio device %d is not usable/enabled", i);
+ continue;
+ }
+
+ GST_DEBUG_OBJECT (obj, "audio device %d looks ok: %s (\"%s\")", i,
+ ai.devnode, ai.name);
+
+ devices = g_list_prepend (devices, g_strdup (ai.devnode));
+ }
+
+ return g_list_reverse (devices);
+}
+
+static GValueArray *
+gst_oss4_property_probe_get_values (GstPropertyProbe * probe,
+ guint prop_id, const GParamSpec * pspec)
+{
+ struct oss_sysinfo si = { {0,}, };
+ GstElementClass *klass;
+ GValueArray *array = NULL;
+ GstObject *obj;
+ GList *devices, *l;
+ int cap_mask, fd = -1;
+
+ if (!g_str_equal (pspec->name, "device")) {
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (probe, prop_id, pspec);
+ return NULL;
+ }
+
+ obj = GST_OBJECT (probe);
+
+ GST_OBJECT_LOCK (obj);
+
+ /* figure out whether the element is a source or sink */
+ klass = GST_ELEMENT_GET_CLASS (GST_ELEMENT (probe));
+ if (GST_IS_OSS4_SINK (probe)) {
+ GST_DEBUG_OBJECT (probe, "probing available output devices");
+ cap_mask = PCM_CAP_OUTPUT;
+ fd = GST_OSS4_SINK (probe)->fd;
+ } else if (GST_IS_OSS4_SOURCE (probe)) {
+ GST_DEBUG_OBJECT (probe, "probing available input devices");
+ cap_mask = PCM_CAP_INPUT;
+ fd = GST_OSS4_SOURCE (probe)->fd;
+ } else if (GST_IS_OSS4_MIXER (probe)) {
+ fd = GST_OSS4_MIXER (probe)->fd;
+ cap_mask = 0;
+ } else {
+ GST_OBJECT_UNLOCK (obj);
+ g_return_val_if_reached (NULL);
+ }
+
+ /* copy fd if it's open, so we can just unconditionally close() later */
+ if (fd != -1)
+ fd = dup (fd);
+
+ /* this will also catch the unlikely case where the above dup() failed */
+ if (fd == -1) {
+ fd = open ("/dev/mixer", O_RDONLY | O_NONBLOCK, 0);
+ if (fd < 0)
+ goto open_failed;
+ else if (!gst_oss4_audio_check_version (GST_OBJECT (probe), fd))
+ goto legacy_oss;
+ }
+
+ if (ioctl (fd, SNDCTL_SYSINFO, &si) == -1)
+ goto no_sysinfo;
+
+ if (cap_mask != 0) {
+ devices =
+ gst_oss4_property_probe_get_audio_devices (obj, fd, &si, cap_mask);
+ } else {
+ devices = gst_oss4_property_probe_get_mixer_devices (obj, fd, &si);
+ }
+
+ if (devices == NULL) {
+ GST_DEBUG_OBJECT (obj, "No devices found");
+ goto done;
+ }
+
+ array = g_value_array_new (1);
+
+ for (l = devices; l != NULL; l = l->next) {
+ GValue val = { 0, };
+
+ g_value_init (&val, G_TYPE_STRING);
+ g_value_take_string (&val, (gchar *) l->data);
+ l->data = NULL;
+ g_value_array_append (array, &val);
+ g_value_unset (&val);
+ }
+
+ GST_OBJECT_UNLOCK (obj);
+
+ g_list_free (devices);
+
+done:
+
+ close (fd);
+
+ return array;
+
+/* ERRORS */
+open_failed:
+ {
+ GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+ GST_WARNING_OBJECT (probe, "Can't open file descriptor to probe "
+ "available devices: %s", g_strerror (errno));
+ return NULL;
+ }
+legacy_oss:
+ {
+ close (fd);
+ GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+ GST_DEBUG_OBJECT (probe, "Legacy OSS (ie. not OSSv4), not supported");
+ return NULL;
+ }
+no_sysinfo:
+ {
+ close (fd);
+ GST_OBJECT_UNLOCK (GST_OBJECT (probe));
+ GST_WARNING_OBJECT (probe, "Can't open file descriptor to probe "
+ "available devices: %s", g_strerror (errno));
+ return NULL;
+ }
+}
+
+static void
+gst_oss4_property_probe_interface_init (GstPropertyProbeInterface * iface)
+{
+ iface->get_properties = gst_oss4_property_probe_get_properties;
+ iface->probe_property = gst_oss4_property_probe_probe_property;
+ iface->needs_probe = gst_oss4_property_probe_needs_probe;
+ iface->get_values = gst_oss4_property_probe_get_values;
+}
+
+void
+gst_oss4_add_property_probe_interface (GType type)
+{
+ static const GInterfaceInfo probe_iface_info = {
+ (GInterfaceInitFunc) gst_oss4_property_probe_interface_init,
+ NULL,
+ NULL,
+ };
+
+ g_type_add_interface_static (type, GST_TYPE_PROPERTY_PROBE,
+ &probe_iface_info);
+}
--- /dev/null
+/* GStreamer OSS4 audio property probe interface implementation
+ * 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.
+ */
+
+#ifndef GST_OSS4_PROPERTY_PROBE_H
+#define GST_OSS4_PROPERTY_PROBE_H
+
+#include <gst/interfaces/propertyprobe.h>
+
+void gst_oss4_add_property_probe_interface (GType type);
+
+gboolean gst_oss4_property_probe_find_device_name (GstObject * obj,
+ int fd,
+ const gchar * device_handle,
+ gchar ** device_name);
+
+#endif /* GST_OSS4_PROPERTY_PROBE_H */
+
+
--- /dev/null
+/* GStreamer OSS4 audio sink
+ * 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.
+ */
+/**
+ * SECTION:element-oss4sink
+ * @short_description: output sound using OSS4
+ *
+ * <refsect2>
+ * <para>
+ * This element lets you output sound using the Open Sound System (OSS)
+ * version 4.
+ * </para>
+ * <para>
+ * Note that you should almost always use generic audio conversion elements
+ * like audioconvert and audioresample in front of an audiosink to make sure
+ * your pipeline works under all circumstances (those conversion elements will
+ * act in passthrough-mode if no conversion is necessary).
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * <programlisting>
+ * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! oss4sink
+ * </programlisting>
+ * will output a sine wave (continuous beep sound) to your sound card (with
+ * a very low volume as precaution).
+ * </para>
+ * <para>
+ * <programlisting>
+ * gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! oss4sink
+ * </programlisting>
+ * will play an Ogg/Vorbis audio file and output it using the Open Sound System
+ * version 4.
+ * </para>
+ * </refsect2>
+ *
+ * Since: 0.10.7
+ */
+
+/* TODO: - add "volume" property for stream volume control and intercept tags
+ * to set stream title
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-audio.h"
+#include "oss4-sink.h"
+#include "oss4-property-probe.h"
+#include "oss4-soundcard.h"
+
+GST_DEBUG_CATEGORY_EXTERN (oss4sink_debug);
+#define GST_CAT_DEFAULT oss4sink_debug
+
+static void gst_oss4_sink_init_interfaces (GType type);
+static void gst_oss4_sink_dispose (GObject * object);
+static void gst_oss4_sink_finalise (GObject * object);
+
+static void gst_oss4_sink_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void gst_oss4_sink_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_oss4_sink_getcaps (GstBaseSink * bsink);
+static gboolean gst_oss4_sink_open (GstAudioSink * asink,
+ gboolean silent_errors);
+static gboolean gst_oss4_sink_open_func (GstAudioSink * asink);
+static gboolean gst_oss4_sink_close (GstAudioSink * asink);
+static gboolean gst_oss4_sink_prepare (GstAudioSink * asink,
+ GstRingBufferSpec * spec);
+static gboolean gst_oss4_sink_unprepare (GstAudioSink * asink);
+static guint gst_oss4_sink_write (GstAudioSink * asink, gpointer data,
+ guint length);
+static guint gst_oss4_sink_delay (GstAudioSink * asink);
+static void gst_oss4_sink_reset (GstAudioSink * asink);
+
+#define DEFAULT_DEVICE NULL
+#define DEFAULT_DEVICE_NAME NULL
+
+enum
+{
+ PROP_0,
+ PROP_DEVICE,
+ PROP_DEVICE_NAME
+};
+
+GST_BOILERPLATE_FULL (GstOss4Sink, gst_oss4_sink, GstAudioSink,
+ GST_TYPE_AUDIO_SINK, gst_oss4_sink_init_interfaces);
+
+static void
+gst_oss4_sink_dispose (GObject * object)
+{
+ GstOss4Sink *osssink = GST_OSS4_SINK (object);
+
+ if (osssink->probed_caps) {
+ gst_caps_unref (osssink->probed_caps);
+ osssink->probed_caps = NULL;
+ }
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_oss4_sink_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+ GstPadTemplate *templ;
+
+ gst_element_class_set_details_simple (element_class,
+ "OSS v4 Audio Sink", "Sink/Audio",
+ "Output to a sound card via OSS version 4",
+ "Tim-Philipp Müller <tim centricular net>");
+
+ templ = gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
+ gst_oss4_audio_get_template_caps ());
+ gst_element_class_add_pad_template (element_class, templ);
+}
+
+static void
+gst_oss4_sink_class_init (GstOss4SinkClass * klass)
+{
+ GstAudioSinkClass *audiosink_class = (GstAudioSinkClass *) klass;
+ GstBaseSinkClass *basesink_class = (GstBaseSinkClass *) klass;
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+
+ gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss4_sink_dispose);
+ gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_oss4_sink_finalise);
+ gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_oss4_sink_get_property);
+ gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_oss4_sink_set_property);
+
+ g_object_class_install_property (gobject_class, PROP_DEVICE,
+ g_param_spec_string ("device", "Device",
+ "OSS4 device (e.g. /dev/oss/hdaudio0/pcm0 or /dev/dspN) "
+ "(NULL = use first available playback device)",
+ DEFAULT_DEVICE, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
+ g_param_spec_string ("device-name", "Device name",
+ "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
+ G_PARAM_READABLE));
+
+ basesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss4_sink_getcaps);
+
+ audiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss4_sink_open_func);
+ audiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss4_sink_close);
+ audiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss4_sink_prepare);
+ audiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss4_sink_unprepare);
+ audiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss4_sink_write);
+ audiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss4_sink_delay);
+ audiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss4_sink_reset);
+}
+
+static void
+gst_oss4_sink_init (GstOss4Sink * osssink, GstOss4SinkClass * klass)
+{
+ const gchar *device;
+
+ device = g_getenv ("AUDIODEV");
+ if (device == NULL)
+ device = DEFAULT_DEVICE;
+ osssink->device = g_strdup (device);
+ osssink->fd = -1;
+ osssink->bytes_per_sample = 0;
+ osssink->probed_caps = NULL;
+ osssink->device_name = NULL;
+}
+
+static void
+gst_oss4_sink_finalise (GObject * object)
+{
+ GstOss4Sink *osssink = GST_OSS4_SINK (object);
+
+ g_free (osssink->device);
+ osssink->device = NULL;
+
+ g_list_free (osssink->property_probe_list);
+ osssink->property_probe_list = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_oss4_sink_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstOss4Sink *oss = GST_OSS4_SINK (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ GST_OBJECT_LOCK (oss);
+ if (oss->fd == -1) {
+ g_free (oss->device);
+ oss->device = g_value_dup_string (value);
+ if (oss->probed_caps) {
+ gst_caps_unref (oss->probed_caps);
+ oss->probed_caps = NULL;
+ }
+ g_free (oss->device_name);
+ oss->device_name = NULL;
+ } else {
+ g_warning ("%s: can't change \"device\" property while audio sink "
+ "is open", GST_OBJECT_NAME (oss));
+ }
+ GST_OBJECT_UNLOCK (oss);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_oss4_sink_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstOss4Sink *oss = GST_OSS4_SINK (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ GST_OBJECT_LOCK (oss);
+ g_value_set_string (value, oss->device);
+ GST_OBJECT_UNLOCK (oss);
+ break;
+ case PROP_DEVICE_NAME:
+ GST_OBJECT_LOCK (oss);
+ if (oss->fd == -1 && oss->device != NULL) {
+ /* If device is set, try to retrieve the name even if we're not open */
+ if (gst_oss4_sink_open (GST_AUDIO_SINK (oss), TRUE)) {
+ g_value_set_string (value, oss->device_name);
+ gst_oss4_sink_close (GST_AUDIO_SINK (oss));
+ } else {
+ g_value_set_string (value, oss->device_name);
+ }
+ } else {
+ g_value_set_string (value, oss->device_name);
+ }
+ GST_OBJECT_UNLOCK (oss);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstCaps *
+gst_oss4_sink_getcaps (GstBaseSink * bsink)
+{
+ GstOss4Sink *oss;
+ GstCaps *caps;
+
+ oss = GST_OSS4_SINK (bsink);
+
+ if (oss->fd == -1) {
+ caps = gst_caps_copy (gst_oss4_audio_get_template_caps ());
+ } else if (oss->probed_caps) {
+ caps = gst_caps_copy (oss->probed_caps);
+ } else {
+ caps = gst_oss4_audio_probe_caps (GST_OBJECT (oss), oss->fd);
+ if (caps != NULL && !gst_caps_is_empty (caps)) {
+ oss->probed_caps = gst_caps_copy (caps);
+ }
+ }
+
+ return caps;
+}
+
+/* note: we must not take the object lock here unless we fix up get_property */
+static gboolean
+gst_oss4_sink_open (GstAudioSink * asink, gboolean silent_errors)
+{
+ GstOss4Sink *oss;
+ gchar *device;
+ int mode;
+
+ oss = GST_OSS4_SINK (asink);
+
+ if (oss->device)
+ device = g_strdup (oss->device);
+ else
+ device = gst_oss4_audio_find_device (GST_OBJECT_CAST (oss));
+
+ /* desperate times, desperate measures */
+ if (device == NULL)
+ device = g_strdup ("/dev/dsp0");
+
+ GST_INFO_OBJECT (oss, "Trying to open OSS4 device '%s'", device);
+
+ /* we open in non-blocking mode even if we don't really want to do writes
+ * non-blocking because we can't be sure that this is really a genuine
+ * OSS4 device with well-behaved drivers etc. We really don't want to
+ * hang forever under any circumstances. */
+ oss->fd = open (device, O_WRONLY | O_NONBLOCK, 0);
+ if (oss->fd == -1) {
+ switch (errno) {
+ case EBUSY:
+ goto busy;
+ case EACCES:
+ goto no_permission;
+ default:
+ goto open_failed;
+ }
+ }
+
+ GST_INFO_OBJECT (oss, "Opened device '%s'", device);
+
+ /* Make sure it's OSS4. If it's old OSS, let osssink handle it */
+ if (!gst_oss4_audio_check_version (GST_OBJECT_CAST (oss), oss->fd))
+ goto legacy_oss;
+
+ /* now remove the non-blocking flag. */
+ mode = fcntl (oss->fd, F_GETFL);
+ mode &= ~O_NONBLOCK;
+ if (fcntl (oss->fd, F_SETFL, mode) < 0) {
+ /* some drivers do no support unsetting the non-blocking flag, try to
+ * close/open the device then. This is racy but we error out properly. */
+ GST_WARNING_OBJECT (oss, "failed to unset O_NONBLOCK (buggy driver?), "
+ "will try to re-open device now");
+ gst_oss4_sink_close (asink);
+ if ((oss->fd = open (device, O_WRONLY, 0)) == -1)
+ goto non_block;
+ }
+
+ oss->open_device = device;
+
+ /* not using ENGINEINFO here because it sometimes returns a different and
+ * less useful name than AUDIOINFO for the same device */
+ if (!gst_oss4_property_probe_find_device_name (GST_OBJECT (oss), oss->fd,
+ oss->open_device, &oss->device_name)) {
+ oss->device_name = NULL;
+ }
+
+ /* list output routings, for informational purposes only so far */
+ {
+ oss_mixer_enuminfo routings = { 0, };
+ guint i;
+
+ if (ioctl (oss->fd, SNDCTL_DSP_GET_PLAYTGT_NAMES, &routings) != -1) {
+ GST_LOG_OBJECT (oss, "%u output routings (static list: %d)",
+ routings.nvalues, !!(routings.version == 0));
+ for (i = 0; i < routings.nvalues; ++i) {
+ GST_LOG_OBJECT (oss, " output routing %d: %s", i,
+ &routings.strings[routings.strindex[i]]);
+ }
+ }
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+busy:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, BUSY,
+ (_("Could not open audio device for playback. "
+ "Device is being used by another application.")), (NULL));
+ }
+ g_free (device);
+ return FALSE;
+ }
+no_permission:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
+ (_("Could not open audio device for playback."
+ "You don't have permission to open the device.")),
+ GST_ERROR_SYSTEM);
+ }
+ g_free (device);
+ return FALSE;
+ }
+open_failed:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
+ (_("Could not open audio device for playback.")), GST_ERROR_SYSTEM);
+ }
+ g_free (device);
+ return FALSE;
+ }
+legacy_oss:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
+ (_("Could not open audio device for playback."
+ "This version of the Open Sound System is not supported by this "
+ "element.")), ("Try the 'osssink' element instead"));
+ }
+ gst_oss4_sink_close (asink);
+ g_free (device);
+ return FALSE;
+ }
+non_block:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
+ ("Unable to set device %s into non-blocking mode: %s",
+ oss->device, g_strerror (errno)));
+ }
+ g_free (device);
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_oss4_sink_open_func (GstAudioSink * asink)
+{
+ return gst_oss4_sink_open (asink, FALSE);
+}
+
+static gboolean
+gst_oss4_sink_close (GstAudioSink * asink)
+{
+ GstOss4Sink *oss = GST_OSS4_SINK (asink);
+
+ if (oss->fd != -1) {
+ GST_DEBUG_OBJECT (oss, "closing device");
+ close (oss->fd);
+ oss->fd = -1;
+ }
+
+ oss->bytes_per_sample = 0;
+ /* we keep the probed caps cached, at least until the device changes */
+
+ g_free (oss->open_device);
+ oss->open_device = NULL;
+
+ g_free (oss->device_name);
+ oss->device_name = NULL;
+
+ if (oss->probed_caps) {
+ gst_caps_unref (oss->probed_caps);
+ oss->probed_caps = NULL;
+ }
+
+ return TRUE;
+}
+
+static gboolean
+gst_oss4_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
+{
+ GstOss4Sink *oss;
+
+ oss = GST_OSS4_SINK (asink);
+
+ if (!gst_oss4_audio_set_format (GST_OBJECT_CAST (oss), oss->fd, spec)) {
+ GST_WARNING_OBJECT (oss, "Couldn't set requested format %" GST_PTR_FORMAT,
+ spec->caps);
+ return FALSE;
+ }
+
+ oss->bytes_per_sample = spec->bytes_per_sample;
+ return TRUE;
+}
+
+static gboolean
+gst_oss4_sink_unprepare (GstAudioSink * asink)
+{
+ /* could do a SNDCTL_DSP_HALT, but the OSS manual recommends a close/open,
+ * since HALT won't properly reset some devices, apparently */
+
+ if (!gst_oss4_sink_close (asink))
+ goto couldnt_close;
+
+ if (!gst_oss4_sink_open_func (asink))
+ goto couldnt_reopen;
+
+ return TRUE;
+
+ /* ERRORS */
+couldnt_close:
+ {
+ GST_DEBUG_OBJECT (asink, "Couldn't close the audio device");
+ return FALSE;
+ }
+couldnt_reopen:
+ {
+ GST_DEBUG_OBJECT (asink, "Couldn't reopen the audio device");
+ return FALSE;
+ }
+}
+
+static guint
+gst_oss4_sink_write (GstAudioSink * asink, gpointer data, guint length)
+{
+ GstOss4Sink *oss;
+ int n;
+
+ oss = GST_OSS4_SINK_CAST (asink);
+
+ n = write (oss->fd, data, length);
+ GST_LOG_OBJECT (asink, "wrote %d/%d samples, %d bytes",
+ n / oss->bytes_per_sample, length / oss->bytes_per_sample, n);
+
+ if (G_UNLIKELY (n < 0)) {
+ switch (errno) {
+ case ENOTSUP:
+ case EACCES:{
+ /* This is the most likely cause, I think */
+ GST_ELEMENT_ERROR (asink, RESOURCE, WRITE,
+ (_("Playback is not supported by this audio device.")),
+ ("write: %s (device: %s) (maybe this is an input-only device?)",
+ g_strerror (errno), oss->open_device));
+ break;
+ }
+ default:{
+ GST_ELEMENT_ERROR (asink, RESOURCE, WRITE,
+ (_("Audio playback error.")),
+ ("write: %s (device: %s)", g_strerror (errno), oss->open_device));
+ break;
+ }
+ }
+ }
+
+ return n;
+}
+
+static guint
+gst_oss4_sink_delay (GstAudioSink * asink)
+{
+ GstOss4Sink *oss;
+ gint delay = -1;
+
+ oss = GST_OSS4_SINK_CAST (asink);
+
+ if (ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay) < 0 || delay < 0) {
+ GST_LOG_OBJECT (oss, "GETODELAY failed");
+ return 0;
+ }
+
+ return delay / oss->bytes_per_sample;
+}
+
+static void
+gst_oss4_sink_reset (GstAudioSink * asink)
+{
+ /* There's nothing we can do here really: OSS can't handle access to the
+ * same device/fd from multiple threads and might deadlock or blow up in
+ * other ways if we try an ioctl SNDCTL_DSP_HALT or similar */
+}
+
+static void
+gst_oss4_sink_init_interfaces (GType type)
+{
+ gst_oss4_add_property_probe_interface (type);
+}
--- /dev/null
+/* GStreamer OSS4 audio sink
+ * 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.
+ */
+
+#ifndef GST_OSS4_SINK_H
+#define GST_OSS4_SINK_H
+
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiosink.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_SINK (gst_oss4_sink_get_type())
+#define GST_OSS4_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_SINK,GstOss4Sink))
+#define GST_OSS4_SINK_CAST(obj) ((GstOss4Sink *)(obj))
+#define GST_OSS4_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_SINK,GstOss4SinkClass))
+#define GST_IS_OSS4_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_SINK))
+#define GST_IS_OSS4_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_SINK))
+
+typedef struct _GstOss4Sink GstOss4Sink;
+typedef struct _GstOss4SinkClass GstOss4SinkClass;
+
+struct _GstOss4Sink {
+ GstAudioSink audio_sink;
+
+ gchar * device; /* NULL if none was set */
+ gchar * open_device; /* the device we opened */
+ gchar * device_name; /* set if the device is open */
+ gint fd; /* -1 if not open */
+ gint bytes_per_sample;
+
+ GstCaps * probed_caps;
+
+ GList * property_probe_list;
+};
+
+struct _GstOss4SinkClass {
+ GstAudioSinkClass audio_sink_class;
+};
+
+GType gst_oss4_sink_get_type (void);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_SINK_H */
+
+
--- /dev/null
+/*
+ * Purpose: The C/C++ header file that defines the OSS API.
+ * Description:
+ * This header file contains all the declarations required to compile OSS
+ * programs. The latest version is always installed together with OSS
+ * use of the latest version is strongly recommended.
+ *
+ * {!notice This header file contains many obsolete definitions
+ * (for compatibility with older applications that still ned them).
+ * Do not use this file as a reference manual of OSS.
+ * Please check the OSS Programmer's guide for descriptions
+ * of the supported API details (http://manuals.opensound.com/developer).}
+ */
+
+#ifndef SOUNDCARD_H
+#define SOUNDCARD_H
+
+#define COPYING40 Copyright (C) 4Front Technologies 2000-2006. Released under the BSD license.
+
+#if defined(__cplusplus)
+#define EXTERNC extern "C"
+#else
+#define EXTERNC extern
+#endif /* EXTERN_C_WRAPPERS */
+
+#define OSS_VERSION 0x040090 // Pre 4.1
+
+#define SOUND_VERSION OSS_VERSION
+#define OPEN_SOUND_SYSTEM
+
+#if defined(__hpux) && !defined(_HPUX_SOURCE)
+# error "-D_HPUX_SOURCE must be used when compiling OSS applications"
+#endif
+
+#ifdef __hpux
+#include <sys/ioctl.h>
+#endif
+
+#ifdef linux
+/* In Linux we need to be prepared for cross compiling */
+#include <linux/ioctl.h>
+#else
+# ifdef __FreeBSD__
+# include <sys/ioccom.h>
+# else
+# include <sys/ioctl.h>
+# endif
+#endif
+
+#ifndef __SIOWR
+#if defined(__hpux) || (defined(_IOWR) && (defined(_AIX) || (!defined(sun) && !defined(sparc) && !defined(__INCioctlh) && !defined(__Lynx__))))
+
+/*
+ * Make sure the ioctl macros are compatible with the ones already used
+ * by this operating system.
+ */
+#define SIOCPARM_MASK IOCPARM_MASK
+#define SIOC_VOID IOC_VOID
+#define SIOC_OUT IOC_OUT
+#define SIOC_IN IOC_IN
+#define SIOC_INOUT IOC_INOUT
+#define __SIOC_SIZE _IOC_SIZE
+#define __SIOC_DIR _IOC_DIR
+#define __SIOC_NONE _IOC_NONE
+#define __SIOC_READ _IOC_READ
+#define __SIOC_WRITE _IOC_WRITE
+#define __SIO _IO
+#define __SIOR _IOR
+#define __SIOW _IOW
+#define __SIOWR _IOWR
+#else
+
+/* #define SIOCTYPE (0xff<<8) */
+#define SIOCPARM_MASK 0x1fff /* parameters must be < 8192 bytes */
+#define SIOC_VOID 0x00000000 /* no parameters */
+#define SIOC_OUT 0x20000000 /* copy out parameters */
+#define SIOC_IN 0x40000000 /* copy in parameters */
+#define SIOC_INOUT (SIOC_IN|SIOC_OUT)
+
+#define __SIO(x,y) ((int)(SIOC_VOID|(x<<8)|y))
+#define __SIOR(x,y,t) ((int)(SIOC_OUT|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y))
+#define __SIOW(x,y,t) ((int)(SIOC_IN|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y))
+#define __SIOWR(x,y,t) ((int)(SIOC_INOUT|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y))
+#define __SIOC_SIZE(x) ((x>>16)&SIOCPARM_MASK)
+#define __SIOC_DIR(x) (x & 0xf0000000)
+#define __SIOC_NONE SIOC_VOID
+#define __SIOC_READ SIOC_OUT
+#define __SIOC_WRITE SIOC_IN
+# endif /* _IOWR */
+#endif /* !__SIOWR */
+
+#define OSS_LONGNAME_SIZE 64
+#define OSS_LABEL_SIZE 16
+#define OSS_DEVNODE_SIZE 32
+typedef char oss_longname_t[OSS_LONGNAME_SIZE];
+typedef char oss_label_t[OSS_LABEL_SIZE];
+typedef char oss_devnode_t[OSS_DEVNODE_SIZE];
+
+#ifndef DISABLE_SEQUENCER
+/*
+ ****************************************************************************
+ * IOCTL Commands for /dev/sequencer and /dev/music (AKA /dev/sequencer2)
+ *
+ * Note that this interface is obsolete and no longer developed. New
+ * applications should use /dev/midi instead.
+ ****************************************************************************/
+#define SNDCTL_SEQ_RESET __SIO ('Q', 0)
+#define SNDCTL_SEQ_SYNC __SIO ('Q', 1)
+#define SNDCTL_SYNTH_INFO __SIOWR('Q', 2, struct synth_info)
+#define SNDCTL_SEQ_CTRLRATE __SIOWR('Q', 3, int) /* Set/get timer resolution (HZ) */
+#define SNDCTL_SEQ_GETOUTCOUNT __SIOR ('Q', 4, int)
+#define SNDCTL_SEQ_GETINCOUNT __SIOR ('Q', 5, int)
+#define SNDCTL_SEQ_PERCMODE __SIOW ('Q', 6, int)
+#define SNDCTL_FM_LOAD_INSTR __SIOW ('Q', 7, struct sbi_instrument) /* Obsolete. Don't use!!!!!! */
+#define SNDCTL_SEQ_TESTMIDI __SIOW ('Q', 8, int)
+#define SNDCTL_SEQ_RESETSAMPLES __SIOW ('Q', 9, int)
+#define SNDCTL_SEQ_NRSYNTHS __SIOR ('Q',10, int)
+#define SNDCTL_SEQ_NRMIDIS __SIOR ('Q',11, int)
+#define SNDCTL_MIDI_INFO __SIOWR('Q',12, struct midi_info) /* OBSOLETE - use SNDCTL_MIDIINFO instead */
+#define SNDCTL_SEQ_THRESHOLD __SIOW ('Q',13, int)
+#define SNDCTL_SYNTH_MEMAVL __SIOWR('Q',14, int) /* in=dev#, out=memsize */
+#define SNDCTL_FM_4OP_ENABLE __SIOW ('Q',15, int) /* in=dev# */
+#define SNDCTL_SEQ_PANIC __SIO ('Q',17)
+#define SNDCTL_SEQ_OUTOFBAND __SIOW ('Q',18, struct seq_event_rec)
+#define SNDCTL_SEQ_GETTIME __SIOR ('Q',19, int)
+#define SNDCTL_SYNTH_ID __SIOWR('Q',20, struct synth_info)
+#define SNDCTL_SYNTH_CONTROL __SIOWR('Q',21, struct synth_control)
+#define SNDCTL_SYNTH_REMOVESAMPLE __SIOWR('Q',22, struct remove_sample) /* Reserved for future use */
+#define SNDCTL_SEQ_TIMING_ENABLE __SIO ('Q', 23) /* Enable incoming MIDI timing messages */
+#define SNDCTL_SEQ_ACTSENSE_ENABLE __SIO ('Q', 24) /* Enable incoming active sensing messages */
+#define SNDCTL_SEQ_RT_ENABLE __SIO ('Q', 25) /* Enable other incoming realtime messages */
+
+typedef struct synth_control
+{
+ int devno; /* Synthesizer # */
+ char data[4000]; /* Device spesific command/data record */
+} synth_control;
+
+typedef struct remove_sample
+{
+ int devno; /* Synthesizer # */
+ int bankno; /* MIDI bank # (0=General MIDI) */
+ int instrno; /* MIDI instrument number */
+} remove_sample;
+
+typedef struct seq_event_rec
+{
+ unsigned char arr[8];
+} seq_event_rec;
+
+#define SNDCTL_TMR_TIMEBASE __SIOWR('T', 1, int)
+#define SNDCTL_TMR_START __SIO ('T', 2)
+#define SNDCTL_TMR_STOP __SIO ('T', 3)
+#define SNDCTL_TMR_CONTINUE __SIO ('T', 4)
+#define SNDCTL_TMR_TEMPO __SIOWR('T', 5, int)
+#define SNDCTL_TMR_SOURCE __SIOWR('T', 6, int)
+# define TMR_INTERNAL 0x00000001
+# define TMR_EXTERNAL 0x00000002
+# define TMR_MODE_MIDI 0x00000010
+# define TMR_MODE_FSK 0x00000020
+# define TMR_MODE_CLS 0x00000040
+# define TMR_MODE_SMPTE 0x00000080
+#define SNDCTL_TMR_METRONOME __SIOW ('T', 7, int)
+#define SNDCTL_TMR_SELECT __SIOW ('T', 8, int)
+
+/*
+ * Sample loading mechanism for internal synthesizers (/dev/sequencer)
+ * (for the .PAT format).
+ */
+
+struct patch_info
+{
+ unsigned short key; /* Use WAVE_PATCH here */
+#define WAVE_PATCH _PATCHKEY(0x04)
+#define GUS_PATCH WAVE_PATCH
+#define WAVEFRONT_PATCH _PATCHKEY(0x06)
+
+ short device_no; /* Synthesizer number */
+ short instr_no; /* Midi pgm# */
+
+ unsigned int mode;
+/*
+ * The least significant byte has the same format than the GUS .PAT
+ * files
+ */
+#define WAVE_16_BITS 0x01 /* bit 0 = 8 or 16 bit wave data. */
+#define WAVE_UNSIGNED 0x02 /* bit 1 = Signed - Unsigned data. */
+#define WAVE_LOOPING 0x04 /* bit 2 = looping enabled-1. */
+#define WAVE_BIDIR_LOOP 0x08 /* bit 3 = Set is bidirectional looping. */
+#define WAVE_LOOP_BACK 0x10 /* bit 4 = Set is looping backward. */
+#define WAVE_SUSTAIN_ON 0x20 /* bit 5 = Turn sustaining on. (Env. pts. 3) */
+#define WAVE_ENVELOPES 0x40 /* bit 6 = Enable envelopes - 1 */
+#define WAVE_FAST_RELEASE 0x80 /* bit 7 = Shut off immediately after note off */
+ /* (use the env_rate/env_offs fields). */
+/* Linux specific bits */
+#define WAVE_VIBRATO 0x00010000 /* The vibrato info is valid */
+#define WAVE_TREMOLO 0x00020000 /* The tremolo info is valid */
+#define WAVE_SCALE 0x00040000 /* The scaling info is valid */
+#define WAVE_FRACTIONS 0x00080000 /* Fraction information is valid */
+/* Reserved bits */
+#define WAVE_ROM 0x40000000 /* For future use */
+#define WAVE_MULAW 0x20000000 /* For future use */
+/* Other bits must be zeroed */
+
+ int len; /* Size of the wave data in bytes */
+ int loop_start, loop_end; /* Byte offsets from the beginning */
+
+/*
+ * The base_freq and base_note fields are used when computing the
+ * playback speed for a note. The base_note defines the tone frequency
+ * which is heard if the sample is played using the base_freq as the
+ * playback speed.
+ *
+ * The low_note and high_note fields define the minimum and maximum note
+ * frequencies for which this sample is valid. It is possible to define
+ * more than one samples for an instrument number at the same time. The
+ * low_note and high_note fields are used to select the most suitable one.
+ *
+ * The fields base_note, high_note and low_note should contain
+ * the note frequency multiplied by 1000. For example value for the
+ * middle A is 440*1000.
+ */
+
+ unsigned int base_freq;
+ unsigned int base_note;
+ unsigned int high_note;
+ unsigned int low_note;
+ int panning; /* -128=left, 127=right */
+ int detuning;
+
+ /* Envelope. Enabled by mode bit WAVE_ENVELOPES */
+ unsigned char env_rate[6]; /* GUS HW ramping rate */
+ unsigned char env_offset[6]; /* 255 == 100% */
+
+ /*
+ * The tremolo, vibrato and scale info are not supported yet.
+ * Enable by setting the mode bits WAVE_TREMOLO, WAVE_VIBRATO or
+ * WAVE_SCALE
+ */
+
+ unsigned char tremolo_sweep;
+ unsigned char tremolo_rate;
+ unsigned char tremolo_depth;
+
+ unsigned char vibrato_sweep;
+ unsigned char vibrato_rate;
+ unsigned char vibrato_depth;
+
+ int scale_frequency;
+ unsigned int scale_factor; /* from 0 to 2048 or 0 to 2 */
+
+ int volume;
+ int fractions;
+ int reserved1;
+ int spare[2];
+ char data[1]; /* The waveform data starts here */
+};
+
+struct sysex_info
+{
+ short key; /* Use SYSEX_PATCH or MAUI_PATCH here */
+#define SYSEX_PATCH _PATCHKEY(0x05)
+#define MAUI_PATCH _PATCHKEY(0x06)
+ short device_no; /* Synthesizer number */
+ int len; /* Size of the sysex data in bytes */
+ unsigned char data[1]; /* Sysex data starts here */
+};
+
+/*
+ * /dev/sequencer input events.
+ *
+ * The data written to the /dev/sequencer is a stream of events. Events
+ * are records of 4 or 8 bytes. The first byte defines the size.
+ * Any number of events can be written with a write call. There
+ * is a set of macros for sending these events. Use these macros if you
+ * want to maximize portability of your program.
+ *
+ * Events SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO. Are also input events.
+ * (All input events are currently 4 bytes long. Be prepared to support
+ * 8 byte events also. If you receive any event having first byte >= 128,
+ * it's a 8 byte event.
+ *
+ * The events are documented at the end of this file.
+ *
+ * Normal events (4 bytes)
+ * There is also a 8 byte version of most of the 4 byte events. The
+ * 8 byte one is recommended.
+ *
+ * NOTE! All 4 byte events are now obsolete. Applications should not write
+ * them. However 4 byte events are still used as inputs from
+ * /dev/sequencer (/dev/music uses only 8 byte ones).
+ */
+#define SEQ_NOTEOFF 0
+#define SEQ_FMNOTEOFF SEQ_NOTEOFF /* Just old name */
+#define SEQ_NOTEON 1
+#define SEQ_FMNOTEON SEQ_NOTEON
+#define SEQ_WAIT TMR_WAIT_ABS
+#define SEQ_PGMCHANGE 3
+#define SEQ_FMPGMCHANGE SEQ_PGMCHANGE
+#define SEQ_SYNCTIMER TMR_START
+#define SEQ_MIDIPUTC 5
+#define SEQ_DRUMON 6 /*** OBSOLETE ***/
+#define SEQ_DRUMOFF 7 /*** OBSOLETE ***/
+#define SEQ_ECHO TMR_ECHO /* For synching programs with output */
+#define SEQ_AFTERTOUCH 9
+#define SEQ_CONTROLLER 10
+#define SEQ_BALANCE 11
+#define SEQ_VOLMODE 12
+
+/************************************
+ * Midi controller numbers *
+ ************************************/
+/*
+ * Controllers 0 to 31 (0x00 to 0x1f) and
+ * 32 to 63 (0x20 to 0x3f) are continuous
+ * controllers.
+ * In the MIDI 1.0 these controllers are sent using
+ * two messages. Controller numbers 0 to 31 are used
+ * to send the MSB and the controller numbers 32 to 63
+ * are for the LSB. Note that just 7 bits are used in MIDI bytes.
+ */
+
+#define CTL_BANK_SELECT 0x00
+#define CTL_MODWHEEL 0x01
+#define CTL_BREATH 0x02
+/* undefined 0x03 */
+#define CTL_FOOT 0x04
+#define CTL_PORTAMENTO_TIME 0x05
+#define CTL_DATA_ENTRY 0x06
+#define CTL_MAIN_VOLUME 0x07
+#define CTL_BALANCE 0x08
+/* undefined 0x09 */
+#define CTL_PAN 0x0a
+#define CTL_EXPRESSION 0x0b
+/* undefined 0x0c */
+/* undefined 0x0d */
+/* undefined 0x0e */
+/* undefined 0x0f */
+#define CTL_GENERAL_PURPOSE1 0x10
+#define CTL_GENERAL_PURPOSE2 0x11
+#define CTL_GENERAL_PURPOSE3 0x12
+#define CTL_GENERAL_PURPOSE4 0x13
+/* undefined 0x14 - 0x1f */
+
+/* undefined 0x20 */
+/* The controller numbers 0x21 to 0x3f are reserved for the */
+/* least significant bytes of the controllers 0x00 to 0x1f. */
+/* These controllers are not recognised by the driver. */
+
+/* Controllers 64 to 69 (0x40 to 0x45) are on/off switches. */
+/* 0=OFF and 127=ON (intermediate values are possible) */
+#define CTL_DAMPER_PEDAL 0x40
+#define CTL_SUSTAIN 0x40 /* Alias */
+#define CTL_HOLD 0x40 /* Alias */
+#define CTL_PORTAMENTO 0x41
+#define CTL_SOSTENUTO 0x42
+#define CTL_SOFT_PEDAL 0x43
+/* undefined 0x44 */
+#define CTL_HOLD2 0x45
+/* undefined 0x46 - 0x4f */
+
+#define CTL_GENERAL_PURPOSE5 0x50
+#define CTL_GENERAL_PURPOSE6 0x51
+#define CTL_GENERAL_PURPOSE7 0x52
+#define CTL_GENERAL_PURPOSE8 0x53
+/* undefined 0x54 - 0x5a */
+#define CTL_EXT_EFF_DEPTH 0x5b
+#define CTL_TREMOLO_DEPTH 0x5c
+#define CTL_CHORUS_DEPTH 0x5d
+#define CTL_DETUNE_DEPTH 0x5e
+#define CTL_CELESTE_DEPTH 0x5e /* Alias for the above one */
+#define CTL_PHASER_DEPTH 0x5f
+#define CTL_DATA_INCREMENT 0x60
+#define CTL_DATA_DECREMENT 0x61
+#define CTL_NONREG_PARM_NUM_LSB 0x62
+#define CTL_NONREG_PARM_NUM_MSB 0x63
+#define CTL_REGIST_PARM_NUM_LSB 0x64
+#define CTL_REGIST_PARM_NUM_MSB 0x65
+/* undefined 0x66 - 0x78 */
+/* reserved 0x79 - 0x7f */
+
+/* Pseudo controllers (not midi compatible) */
+#define CTRL_PITCH_BENDER 255
+#define CTRL_PITCH_BENDER_RANGE 254
+#define CTRL_EXPRESSION 253 /* Obsolete */
+#define CTRL_MAIN_VOLUME 252 /* Obsolete */
+
+/*
+ * Volume mode defines how volumes are used
+ */
+
+#define VOL_METHOD_ADAGIO 1
+#define VOL_METHOD_LINEAR 2
+
+/*
+ * Note! SEQ_WAIT, SEQ_MIDIPUTC and SEQ_ECHO are used also as
+ * input events.
+ */
+
+/*
+ * Event codes 0xf0 to 0xfc are reserved for future extensions.
+ */
+
+#define SEQ_FULLSIZE 0xfd /* Long events */
+/*
+ * SEQ_FULLSIZE events are used for loading patches/samples to the
+ * synthesizer devices. These events are passed directly to the driver
+ * of the associated synthesizer device. There is no limit to the size
+ * of the extended events. These events are not queued but executed
+ * immediately when the write() is called (execution can take several
+ * seconds of time).
+ *
+ * When a SEQ_FULLSIZE message is written to the device, it must
+ * be written using exactly one write() call. Other events cannot
+ * be mixed to the same write.
+ *
+ * For FM synths (YM3812/OPL3) use struct sbi_instrument and write it to the
+ * /dev/sequencer. Don't write other data together with the instrument structure
+ * Set the key field of the structure to FM_PATCH. The device field is used to
+ * route the patch to the corresponding device.
+ *
+ * For wave table use struct patch_info. Initialize the key field
+ * to WAVE_PATCH.
+ */
+#define SEQ_PRIVATE 0xfe /* Low level HW dependent events (8 bytes) */
+#define SEQ_EXTENDED 0xff /* Extended events (8 bytes) OBSOLETE */
+
+/*
+ * Record for FM patches
+ */
+
+typedef unsigned char sbi_instr_data[32];
+
+struct sbi_instrument
+{
+ unsigned short key; /* FM_PATCH or OPL3_PATCH */
+#define FM_PATCH _PATCHKEY(0x01)
+#define OPL3_PATCH _PATCHKEY(0x03)
+ short device; /* Synth# (0-4) */
+ int channel; /* Program# to be initialized */
+ sbi_instr_data operators; /* Register settings for operator cells (.SBI format) */
+};
+
+struct synth_info
+{ /* Read only */
+ char name[30];
+ int device; /* 0-N. INITIALIZE BEFORE CALLING */
+ int synth_type;
+#define SYNTH_TYPE_FM 0
+#define SYNTH_TYPE_SAMPLE 1
+#define SYNTH_TYPE_MIDI 2 /* Midi interface */
+
+ int synth_subtype;
+#define FM_TYPE_ADLIB 0x00
+#define FM_TYPE_OPL3 0x01
+#define MIDI_TYPE_MPU401 0x401
+
+#define SAMPLE_TYPE_BASIC 0x10
+#define SAMPLE_TYPE_GUS SAMPLE_TYPE_BASIC
+#define SAMPLE_TYPE_WAVEFRONT 0x11
+
+ int perc_mode; /* No longer supported */
+ int nr_voices;
+ int nr_drums; /* Obsolete field */
+ int instr_bank_size;
+ unsigned int capabilities;
+#define SYNTH_CAP_PERCMODE 0x00000001 /* No longer used */
+#define SYNTH_CAP_OPL3 0x00000002 /* Set if OPL3 supported */
+#define SYNTH_CAP_INPUT 0x00000004 /* Input (MIDI) device */
+ int dummies[19]; /* Reserve space */
+};
+
+struct sound_timer_info
+{
+ char name[32];
+ int caps;
+};
+
+struct midi_info /* OBSOLETE */
+{
+ char name[30];
+ int device; /* 0-N. INITIALIZE BEFORE CALLING */
+ unsigned int capabilities; /* To be defined later */
+ int dev_type;
+ int dummies[18]; /* Reserve space */
+};
+
+/*
+ * Level 2 event types for /dev/sequencer
+ */
+
+/*
+ * The 4 most significant bits of byte 0 specify the class of
+ * the event:
+ *
+ * 0x8X = system level events,
+ * 0x9X = device/port specific events, event[1] = device/port,
+ * The last 4 bits give the subtype:
+ * 0x02 = Channel event (event[3] = chn).
+ * 0x01 = note event (event[4] = note).
+ * (0x01 is not used alone but always with bit 0x02).
+ * event[2] = MIDI message code (0x80=note off etc.)
+ *
+ */
+
+#define EV_SEQ_LOCAL 0x80
+#define EV_TIMING 0x81
+#define EV_CHN_COMMON 0x92
+#define EV_CHN_VOICE 0x93
+#define EV_SYSEX 0x94
+#define EV_SYSTEM 0x95 /* MIDI system and real time messages (input only) */
+/*
+ * Event types 200 to 220 are reserved for application use.
+ * These numbers will not be used by the driver.
+ */
+
+/*
+ * Events for event type EV_CHN_VOICE
+ */
+
+#define MIDI_NOTEOFF 0x80
+#define MIDI_NOTEON 0x90
+#define MIDI_KEY_PRESSURE 0xA0
+
+/*
+ * Events for event type EV_CHN_COMMON
+ */
+
+#define MIDI_CTL_CHANGE 0xB0
+#define MIDI_PGM_CHANGE 0xC0
+#define MIDI_CHN_PRESSURE 0xD0
+#define MIDI_PITCH_BEND 0xE0
+
+#define MIDI_SYSTEM_PREFIX 0xF0
+
+/*
+ * Timer event types
+ */
+#define TMR_WAIT_REL 1 /* Time relative to the prev time */
+#define TMR_WAIT_ABS 2 /* Absolute time since TMR_START */
+#define TMR_STOP 3
+#define TMR_START 4
+#define TMR_CONTINUE 5
+#define TMR_TEMPO 6
+#define TMR_ECHO 8
+#define TMR_CLOCK 9 /* MIDI clock */
+#define TMR_SPP 10 /* Song position pointer */
+#define TMR_TIMESIG 11 /* Time signature */
+
+/*
+ * Local event types
+ */
+#define LOCL_STARTAUDIO 1
+#define LOCL_STARTAUDIO2 2
+#define LOCL_STARTAUDIO3 3
+#define LOCL_STARTAUDIO4 4
+
+#if (!defined(__KERNEL__) && !defined(KERNEL) && !defined(INKERNEL) && !defined(_KERNEL)) || defined(USE_SEQ_MACROS)
+/*
+ * Some convenience macros to simplify programming of the
+ * /dev/sequencer interface
+ *
+ * These macros define the API which should be used when possible.
+ */
+#define SEQ_DECLAREBUF() SEQ_USE_EXTBUF()
+
+void seqbuf_dump (void); /* This function must be provided by programs */
+
+EXTERNC int OSS_init (int seqfd, int buflen);
+EXTERNC void OSS_seqbuf_dump (int fd, unsigned char *buf, int buflen);
+EXTERNC void OSS_seq_advbuf (int len, int fd, unsigned char *buf, int buflen);
+EXTERNC void OSS_seq_needbuf (int len, int fd, unsigned char *buf,
+ int buflen);
+EXTERNC void OSS_patch_caching (int dev, int chn, int patch, int fd,
+ unsigned char *buf, int buflen);
+EXTERNC void OSS_drum_caching (int dev, int chn, int patch, int fd,
+ unsigned char *buf, int buflen);
+EXTERNC void OSS_write_patch (int fd, unsigned char *buf, int len);
+EXTERNC int OSS_write_patch2 (int fd, unsigned char *buf, int len);
+
+#define SEQ_PM_DEFINES int __foo_bar___
+#ifdef OSSLIB
+# define SEQ_USE_EXTBUF() \
+ EXTERNC unsigned char *_seqbuf; \
+ EXTERNC int _seqbuflen;EXTERNC int _seqbufptr
+# define SEQ_DEFINEBUF(len) SEQ_USE_EXTBUF();static int _requested_seqbuflen=len
+# define _SEQ_ADVBUF(len) OSS_seq_advbuf(len, seqfd, _seqbuf, _seqbuflen)
+# define _SEQ_NEEDBUF(len) OSS_seq_needbuf(len, seqfd, _seqbuf, _seqbuflen)
+# define SEQ_DUMPBUF() OSS_seqbuf_dump(seqfd, _seqbuf, _seqbuflen)
+
+# define SEQ_LOAD_GMINSTR(dev, instr) \
+ OSS_patch_caching(dev, -1, instr, seqfd, _seqbuf, _seqbuflen)
+# define SEQ_LOAD_GMDRUM(dev, drum) \
+ OSS_drum_caching(dev, -1, drum, seqfd, _seqbuf, _seqbuflen)
+#else /* !OSSLIB */
+
+# define SEQ_LOAD_GMINSTR(dev, instr)
+# define SEQ_LOAD_GMDRUM(dev, drum)
+
+# define SEQ_USE_EXTBUF() \
+ EXTERNC unsigned char _seqbuf[]; \
+ EXTERNC int _seqbuflen;EXTERNC int _seqbufptr
+
+#ifndef USE_SIMPLE_MACROS
+/* Sample seqbuf_dump() implementation:
+ *
+ * SEQ_DEFINEBUF (2048); -- Defines a buffer for 2048 bytes
+ *
+ * int seqfd; -- The file descriptor for /dev/sequencer.
+ *
+ * void
+ * seqbuf_dump ()
+ * {
+ * if (_seqbufptr)
+ * if (write (seqfd, _seqbuf, _seqbufptr) == -1)
+ * {
+ * perror ("write /dev/sequencer");
+ * exit (-1);
+ * }
+ * _seqbufptr = 0;
+ * }
+ */
+
+#define SEQ_DEFINEBUF(len) \
+ unsigned char _seqbuf[len]; int _seqbuflen = len;int _seqbufptr = 0
+#define _SEQ_NEEDBUF(len) \
+ if ((_seqbufptr+(len)) > _seqbuflen) seqbuf_dump()
+#define _SEQ_ADVBUF(len) _seqbufptr += len
+#define SEQ_DUMPBUF seqbuf_dump
+#else
+/*
+ * This variation of the sequencer macros is used just to format one event
+ * using fixed buffer.
+ *
+ * The program using the macro library must define the following macros before
+ * using this library.
+ *
+ * #define _seqbuf name of the buffer (unsigned char[])
+ * #define _SEQ_ADVBUF(len) If the applic needs to know the exact
+ * size of the event, this macro can be used.
+ * Otherwise this must be defined as empty.
+ * #define _seqbufptr Define the name of index variable or 0 if
+ * not required.
+ */
+#define _SEQ_NEEDBUF(len) /* empty */
+#endif
+#endif /* !OSSLIB */
+
+#define SEQ_VOLUME_MODE(dev, mode) \
+ {_SEQ_NEEDBUF(8);\
+ _seqbuf[_seqbufptr] = SEQ_EXTENDED;\
+ _seqbuf[_seqbufptr+1] = SEQ_VOLMODE;\
+ _seqbuf[_seqbufptr+2] = (dev);\
+ _seqbuf[_seqbufptr+3] = (mode);\
+ _seqbuf[_seqbufptr+4] = 0;\
+ _seqbuf[_seqbufptr+5] = 0;\
+ _seqbuf[_seqbufptr+6] = 0;\
+ _seqbuf[_seqbufptr+7] = 0;\
+ _SEQ_ADVBUF(8);}
+
+/*
+ * Midi voice messages
+ */
+
+#define _CHN_VOICE(dev, event, chn, note, parm) \
+ {_SEQ_NEEDBUF(8);\
+ _seqbuf[_seqbufptr] = EV_CHN_VOICE;\
+ _seqbuf[_seqbufptr+1] = (dev);\
+ _seqbuf[_seqbufptr+2] = (event);\
+ _seqbuf[_seqbufptr+3] = (chn);\
+ _seqbuf[_seqbufptr+4] = (note);\
+ _seqbuf[_seqbufptr+5] = (parm);\
+ _seqbuf[_seqbufptr+6] = (0);\
+ _seqbuf[_seqbufptr+7] = 0;\
+ _SEQ_ADVBUF(8);}
+
+#define SEQ_START_NOTE(dev, chn, note, vol) \
+ _CHN_VOICE(dev, MIDI_NOTEON, chn, note, vol)
+
+#define SEQ_STOP_NOTE(dev, chn, note, vol) \
+ _CHN_VOICE(dev, MIDI_NOTEOFF, chn, note, vol)
+
+#define SEQ_KEY_PRESSURE(dev, chn, note, pressure) \
+ _CHN_VOICE(dev, MIDI_KEY_PRESSURE, chn, note, pressure)
+
+/*
+ * Midi channel messages
+ */
+
+#define _CHN_COMMON(dev, event, chn, p1, p2, w14) \
+ {_SEQ_NEEDBUF(8);\
+ _seqbuf[_seqbufptr] = EV_CHN_COMMON;\
+ _seqbuf[_seqbufptr+1] = (dev);\
+ _seqbuf[_seqbufptr+2] = (event);\
+ _seqbuf[_seqbufptr+3] = (chn);\
+ _seqbuf[_seqbufptr+4] = (p1);\
+ _seqbuf[_seqbufptr+5] = (p2);\
+ *(short *)&_seqbuf[_seqbufptr+6] = (w14);\
+ _SEQ_ADVBUF(8);}
+/*
+ * SEQ_SYSEX permits sending of sysex messages. (It may look that it permits
+ * sending any MIDI bytes but it's absolutely not possible. Trying to do
+ * so _will_ cause problems with MPU401 intelligent mode).
+ *
+ * Sysex messages are sent in blocks of 1 to 6 bytes. Longer messages must be
+ * sent by calling SEQ_SYSEX() several times (there must be no other events
+ * between them). First sysex fragment must have 0xf0 in the first byte
+ * and the last byte (buf[len-1] of the last fragment must be 0xf7. No byte
+ * between these sysex start and end markers cannot be larger than 0x7f. Also
+ * lengths of each fragments (except the last one) must be 6.
+ *
+ * Breaking the above rules may work with some MIDI ports but is likely to
+ * cause fatal problems with some other devices (such as MPU401).
+ */
+#define SEQ_SYSEX(dev, buf, len) \
+ {int ii, ll=(len); \
+ unsigned char *bufp=buf;\
+ if (ll>6)ll=6;\
+ _SEQ_NEEDBUF(8);\
+ _seqbuf[_seqbufptr] = EV_SYSEX;\
+ _seqbuf[_seqbufptr+1] = (dev);\
+ for(ii=0;ii<ll;ii++)\
+ _seqbuf[_seqbufptr+ii+2] = bufp[ii];\
+ for(ii=ll;ii<6;ii++)\
+ _seqbuf[_seqbufptr+ii+2] = 0xff;\
+ _SEQ_ADVBUF(8);}
+
+#define SEQ_CHN_PRESSURE(dev, chn, pressure) \
+ _CHN_COMMON(dev, MIDI_CHN_PRESSURE, chn, pressure, 0, 0)
+
+#define SEQ_SET_PATCH SEQ_PGM_CHANGE
+#ifdef OSSLIB
+# define SEQ_PGM_CHANGE(dev, chn, patch) \
+ {OSS_patch_caching(dev, chn, patch, seqfd, _seqbuf, _seqbuflen); \
+ _CHN_COMMON(dev, MIDI_PGM_CHANGE, chn, patch, 0, 0);}
+#else
+# define SEQ_PGM_CHANGE(dev, chn, patch) \
+ _CHN_COMMON(dev, MIDI_PGM_CHANGE, chn, patch, 0, 0)
+#endif
+
+#define SEQ_CONTROL(dev, chn, controller, value) \
+ _CHN_COMMON(dev, MIDI_CTL_CHANGE, chn, controller, 0, value)
+
+#define SEQ_BENDER(dev, chn, value) \
+ _CHN_COMMON(dev, MIDI_PITCH_BEND, chn, 0, 0, value)
+
+#define SEQ_V2_X_CONTROL(dev, voice, controller, value) \
+ {_SEQ_NEEDBUF(8);\
+ _seqbuf[_seqbufptr] = SEQ_EXTENDED;\
+ _seqbuf[_seqbufptr+1] = SEQ_CONTROLLER;\
+ _seqbuf[_seqbufptr+2] = (dev);\
+ _seqbuf[_seqbufptr+3] = (voice);\
+ _seqbuf[_seqbufptr+4] = (controller);\
+ _seqbuf[_seqbufptr+5] = ((value)&0xff);\
+ _seqbuf[_seqbufptr+6] = ((value>>8)&0xff);\
+ _seqbuf[_seqbufptr+7] = 0;\
+ _SEQ_ADVBUF(8);}
+/*
+ * The following 5 macros are incorrectly implemented and obsolete.
+ * Use SEQ_BENDER and SEQ_CONTROL (with proper controller) instead.
+ */
+#define SEQ_PITCHBEND(dev, voice, value) \
+ SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER, value)
+#define SEQ_BENDER_RANGE(dev, voice, value) \
+ SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER_RANGE, value)
+#define SEQ_EXPRESSION(dev, voice, value) \
+ SEQ_CONTROL(dev, voice, CTL_EXPRESSION, value*128)
+#define SEQ_MAIN_VOLUME(dev, voice, value) \
+ SEQ_CONTROL(dev, voice, CTL_MAIN_VOLUME, (value*16383)/100)
+#define SEQ_PANNING(dev, voice, pos) \
+ SEQ_CONTROL(dev, voice, CTL_PAN, (pos+128) / 2)
+
+/*
+ * Timing and syncronization macros
+ */
+
+#define _TIMER_EVENT(ev, parm) {_SEQ_NEEDBUF(8);\
+ _seqbuf[_seqbufptr+0] = EV_TIMING; \
+ _seqbuf[_seqbufptr+1] = (ev); \
+ _seqbuf[_seqbufptr+2] = 0;\
+ _seqbuf[_seqbufptr+3] = 0;\
+ *(unsigned int *)&_seqbuf[_seqbufptr+4] = (parm); \
+ _SEQ_ADVBUF(8);}
+
+#define SEQ_START_TIMER() _TIMER_EVENT(TMR_START, 0)
+#define SEQ_STOP_TIMER() _TIMER_EVENT(TMR_STOP, 0)
+#define SEQ_CONTINUE_TIMER() _TIMER_EVENT(TMR_CONTINUE, 0)
+#define SEQ_WAIT_TIME(ticks) _TIMER_EVENT(TMR_WAIT_ABS, ticks)
+#define SEQ_DELTA_TIME(ticks) _TIMER_EVENT(TMR_WAIT_REL, ticks)
+#define SEQ_ECHO_BACK(key) _TIMER_EVENT(TMR_ECHO, key)
+#define SEQ_SET_TEMPO(value) _TIMER_EVENT(TMR_TEMPO, value)
+#define SEQ_SONGPOS(pos) _TIMER_EVENT(TMR_SPP, pos)
+#define SEQ_TIME_SIGNATURE(sig) _TIMER_EVENT(TMR_TIMESIG, sig)
+
+/*
+ * Local control events
+ */
+
+#define _LOCAL_EVENT(ev, parm) {_SEQ_NEEDBUF(8);\
+ _seqbuf[_seqbufptr+0] = EV_SEQ_LOCAL; \
+ _seqbuf[_seqbufptr+1] = (ev); \
+ _seqbuf[_seqbufptr+2] = 0;\
+ _seqbuf[_seqbufptr+3] = 0;\
+ *(unsigned int *)&_seqbuf[_seqbufptr+4] = (parm); \
+ _SEQ_ADVBUF(8);}
+
+#define SEQ_PLAYAUDIO(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO, devmask)
+#define SEQ_PLAYAUDIO2(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO2, devmask)
+#define SEQ_PLAYAUDIO3(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO3, devmask)
+#define SEQ_PLAYAUDIO4(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO4, devmask)
+/*
+ * Events for the level 1 interface only
+ */
+
+#define SEQ_MIDIOUT(device, byte) {_SEQ_NEEDBUF(4);\
+ _seqbuf[_seqbufptr] = SEQ_MIDIPUTC;\
+ _seqbuf[_seqbufptr+1] = (byte);\
+ _seqbuf[_seqbufptr+2] = (device);\
+ _seqbuf[_seqbufptr+3] = 0;\
+ _SEQ_ADVBUF(4);}
+
+/*
+ * Patch loading.
+ */
+#ifdef OSSLIB
+# define SEQ_WRPATCH(patchx, len) \
+ OSS_write_patch(seqfd, (char*)(patchx), len)
+# define SEQ_WRPATCH2(patchx, len) \
+ OSS_write_patch2(seqfd, (char*)(patchx), len)
+#else
+# define SEQ_WRPATCH(patchx, len) \
+ {if (_seqbufptr) SEQ_DUMPBUF();\
+ if (write(seqfd, (char*)(patchx), len)==-1) \
+ perror("Write patch: /dev/sequencer");}
+# define SEQ_WRPATCH2(patchx, len) \
+ (SEQ_DUMPBUF(), write(seqfd, (char*)(patchx), len))
+#endif
+
+#endif
+#endif /* ifndef DISABLE_SEQUENCER */
+
+/*
+ ****************************************************************************
+ * ioctl commands for the /dev/midi##
+ ****************************************************************************/
+#define SNDCTL_MIDI_PRETIME __SIOWR('m', 0, int)
+
+#if 0
+/*
+ * The SNDCTL_MIDI_MPUMODE and SNDCTL_MIDI_MPUCMD calls
+ * are completely obsolete. The hardware device (MPU-401 "intelligent mode"
+ * and compatibles) has disappeared from the market 10 years ago so there
+ * is no need for this stuff. The MPU-401 "UART" mode devices don't support
+ * this stuff.
+ */
+typedef struct
+{
+ unsigned char cmd;
+ char nr_args, nr_returns;
+ unsigned char data[30];
+} mpu_command_rec;
+
+#define SNDCTL_MIDI_MPUMODE __SIOWR('m', 1, int)
+#define SNDCTL_MIDI_MPUCMD __SIOWR('m', 2, mpu_command_rec)
+#endif
+
+/*
+ * SNDCTL_MIDI_MTCINPUT turns on a mode where OSS automatically inserts
+ * MTC quarter frame messages (F1 xx) to the input.
+ * The argument is the MTC mode:
+ *
+ * -1 = Turn MTC messages OFF (default)
+ * 24 = 24 FPS
+ * 25 = 25 FPS
+ * 29 = 30 FPS drop frame
+ * 30 = 30 FPS
+ *
+ * Note that 25 FPS mode is probably the only mode that is supported. Other
+ * modes may be supported in the future versions of OSS, 25 FPS is handy
+ * because it generates 25*4=100 quarter frame messages per second which
+ * matches the usual 100 HZ system timer rate).
+ *
+ * The quarter frame timer will be reset to 0:00:00:00.0 at the moment this
+ * ioctl is made.
+ */
+#define SNDCTL_MIDI_MTCINPUT __SIOWR('m', 3, int)
+
+/*
+ * MTC/SMPTE time code record (for future use)
+ */
+typedef struct
+{
+ unsigned char hours, minutes, seconds, frames, qframes;
+ char direction;
+#define MTC_DIR_STOPPED 0
+#define MTC_DIR_FORWARD 1
+#define MTC_DIR_BACKWARD -1
+ unsigned char time_code_type;
+ unsigned int flags;
+} oss_mtc_data_t;
+
+#define SNDCTL_MIDI_SETMODE __SIOWR('m', 6, int)
+# define MIDI_MODE_TRADITIONAL 0
+# define MIDI_MODE_TIMED 1 /* Input times are in MIDI ticks */
+# define MIDI_MODE_TIMED_ABS 2 /* Input times are absolute (usecs) */
+
+/*
+ * Packet header for MIDI_MODE_TIMED and MIDI_MODE_TIMED_ABS
+ */
+typedef unsigned long long oss_midi_time_t; /* Variable type for MIDI time (clock ticks) */
+
+typedef struct
+{
+ int magic; /* Initialize to MIDI_HDR_MAGIC */
+#define MIDI_HDR_MAGIC -1
+ unsigned short event_type;
+#define MIDI_EV_WRITE 0 /* Write or read (with payload) */
+#define MIDI_EV_TEMPO 1
+#define MIDI_EV_ECHO 2
+#define MIDI_EV_START 3
+#define MIDI_EV_STOP 4
+#define MIDI_EV_CONTINUE 5
+#define MIDI_EV_XPRESSWRITE 6
+#define MIDI_EV_TIMEBASE 7
+#define MIDI_EV_DEVCTL 8 /* Device control read/write */
+ unsigned short options;
+#define MIDI_OPT_NONE 0x0000
+#define MIDI_OPT_TIMED 0x0001
+#define MIDI_OPT_CONTINUATION 0x0002
+#define MIDI_OPT_USECTIME 0x0004 /* Time is absolute (in usecs) */
+#define MIDI_OPT_BUSY 0x0008 /* Reserved for internal use */
+ oss_midi_time_t time;
+ int parm;
+ int filler[3]; /* Fur future expansion - init to zeros */
+} midi_packet_header_t;
+/*
+ * MIDI_PAYLOAD_SIZE is the maximum size of one MIDI input chunk. It must be
+ * less (or equal) than 1024 which is the read size recommended in the
+ * documentation. TODO: Explain this better.
+ */
+#define MIDI_PAYLOAD_SIZE 1000
+
+typedef struct
+{
+ midi_packet_header_t hdr;
+ unsigned char payload[MIDI_PAYLOAD_SIZE];
+} midi_packet_t;
+
+#define SNDCTL_MIDI_TIMEBASE __SIOWR('m', 7, int)
+#define SNDCTL_MIDI_TEMPO __SIOWR('m', 8, int)
+/*
+ * User land MIDI servers (synths) can use SNDCTL_MIDI_SET_LATENCY
+ * to request MIDI events to be sent to them in advance. The parameter
+ * (in microseconds) tells how much before the events are submitted.
+ *
+ * This feature is only valid for loopback devices and possibly some other
+ * types of virtual devices.
+ */
+#define SNDCTL_MIDI_SET_LATENCY __SIOW ('m', 9, int)
+/*
+ ****************************************************************************
+ * IOCTL commands for /dev/dsp
+ ****************************************************************************/
+
+#define SNDCTL_DSP_HALT __SIO ('P', 0)
+#define SNDCTL_DSP_RESET SNDCTL_DSP_HALT /* Old name */
+#define SNDCTL_DSP_SYNC __SIO ('P', 1)
+#define SNDCTL_DSP_SPEED __SIOWR('P', 2, int)
+
+/* SNDCTL_DSP_STEREO is obsolete - use SNDCTL_DSP_CHANNELS instead */
+#define SNDCTL_DSP_STEREO __SIOWR('P', 3, int)
+/* SNDCTL_DSP_STEREO is obsolete - use SNDCTL_DSP_CHANNELS instead */
+
+#define SNDCTL_DSP_GETBLKSIZE __SIOWR('P', 4, int)
+#define SNDCTL_DSP_SAMPLESIZE SNDCTL_DSP_SETFMT
+#define SNDCTL_DSP_CHANNELS __SIOWR('P', 6, int)
+#define SNDCTL_DSP_POST __SIO ('P', 8)
+#define SNDCTL_DSP_SUBDIVIDE __SIOWR('P', 9, int)
+#define SNDCTL_DSP_SETFRAGMENT __SIOWR('P',10, int)
+
+/* Audio data formats (Note! U8=8 and S16_LE=16 for compatibility) */
+#define SNDCTL_DSP_GETFMTS __SIOR ('P',11, int) /* Returns a mask */
+#define SNDCTL_DSP_SETFMT __SIOWR('P',5, int) /* Selects ONE fmt */
+# define AFMT_QUERY 0x00000000 /* Return current fmt */
+# define AFMT_MU_LAW 0x00000001
+# define AFMT_A_LAW 0x00000002
+# define AFMT_IMA_ADPCM 0x00000004
+# define AFMT_U8 0x00000008
+# define AFMT_S16_LE 0x00000010 /* Little endian signed 16 */
+# define AFMT_S16_BE 0x00000020 /* Big endian signed 16 */
+# define AFMT_S8 0x00000040
+# define AFMT_U16_LE 0x00000080 /* Little endian U16 */
+# define AFMT_U16_BE 0x00000100 /* Big endian U16 */
+# define AFMT_MPEG 0x00000200 /* MPEG (2) audio */
+
+/* AC3 _compressed_ bitstreams (See Programmer's Guide for details). */
+# define AFMT_AC3 0x00000400
+/* Ogg Vorbis _compressed_ bit streams */
+# define AFMT_VORBIS 0x00000800
+
+/* 32 bit formats (MSB aligned) formats */
+# define AFMT_S32_LE 0x00001000
+# define AFMT_S32_BE 0x00002000
+
+/* Reserved for _native_ endian double precision IEEE floating point */
+# define AFMT_FLOAT 0x00004000
+
+/* 24 bit formats (LSB aligned in 32 bit word) formats */
+# define AFMT_S24_LE 0x00008000
+# define AFMT_S24_BE 0x00010000
+
+/*
+ * S/PDIF raw format. In this format the S/PDIF frames (including all
+ * control and user bits) are included in the data stream. Each sample
+ * is stored in a 32 bit frame (see IEC-958 for more info). This format
+ * is supported by very few devices and it's only usable for purposes
+ * where full access to the control/user bits is required (real time control).
+ */
+# define AFMT_SPDIF_RAW 0x00020000
+
+/* 24 bit packed (3 byte) little endian format (USB compatibility) */
+# define AFMT_S24_PACKED 0x00040000
+
+
+/*
+ * Some big endian/little endian handling macros (native endian and opposite
+ * endian formats). The usage of these macros is described in the OSS
+ * Programmer's Manual.
+ */
+
+#if defined(_AIX) || defined(AIX) || defined(sparc) || defined(__hppa) || defined(PPC) || defined(__powerpc__) && !defined(i386) && !defined(__i386) && !defined(__i386__)
+
+/* Big endian machines */
+# define _PATCHKEY(id) (0xfd00|id)
+# define AFMT_S16_NE AFMT_S16_BE
+# define AFMT_U16_NE AFMT_U16_BE
+# define AFMT_S32_NE AFMT_S32_BE
+# define AFMT_S24_NE AFMT_S24_BE
+# define AFMT_S16_OE AFMT_S16_LE
+# define AFMT_S32_OE AFMT_S32_LE
+# define AFMT_S24_OE AFMT_S24_LE
+#else
+# define _PATCHKEY(id) ((id<<8)|0xfd)
+# define AFMT_S16_NE AFMT_S16_LE
+# define AFMT_U16_NE AFMT_U16_LE
+# define AFMT_S32_NE AFMT_S32_LE
+# define AFMT_S24_NE AFMT_S24_LE
+# define AFMT_S16_OE AFMT_S16_BE
+# define AFMT_S32_OE AFMT_S32_BE
+# define AFMT_S24_OE AFMT_S24_BE
+#endif
+/*
+ * Buffer status queries.
+ */
+typedef struct audio_buf_info
+{
+ int fragments; /* # of available fragments (partially usend ones not counted) */
+ int fragstotal; /* Total # of fragments allocated */
+ int fragsize; /* Size of a fragment in bytes */
+ int bytes; /* Available space in bytes (includes partially used fragments) */
+ /* Note! 'bytes' could be more than fragments*fragsize */
+} audio_buf_info;
+
+#define SNDCTL_DSP_GETOSPACE __SIOR ('P',12, audio_buf_info)
+#define SNDCTL_DSP_GETISPACE __SIOR ('P',13, audio_buf_info)
+#define SNDCTL_DSP_GETCAPS __SIOR ('P',15, int)
+# define PCM_CAP_REVISION 0x000000ff /* Bits for revision level (0 to 255) */
+# define PCM_CAP_DUPLEX 0x00000100 /* Full duplex record/playback */
+# define PCM_CAP_REALTIME 0x00000200 /* Not in use */
+# define PCM_CAP_BATCH 0x00000400 /* Device has some kind of */
+ /* internal buffers which may */
+ /* cause some delays and */
+ /* decrease precision of timing */
+# define PCM_CAP_COPROC 0x00000800 /* Has a coprocessor */
+ /* Sometimes it's a DSP */
+ /* but usually not */
+# define PCM_CAP_TRIGGER 0x00001000 /* Supports SETTRIGGER */
+# define PCM_CAP_MMAP 0x00002000 /* Supports mmap() */
+# define PCM_CAP_MULTI 0x00004000 /* Supports multiple open */
+# define PCM_CAP_BIND 0x00008000 /* Supports binding to front/rear/center/lfe */
+# define PCM_CAP_INPUT 0x00010000 /* Supports recording */
+# define PCM_CAP_OUTPUT 0x00020000 /* Supports playback */
+# define PCM_CAP_VIRTUAL 0x00040000 /* Virtual device */
+/* 0x00040000 and 0x00080000 reserved for future use */
+
+/* Analog/digital control capabilities */
+# define PCM_CAP_ANALOGOUT 0x00100000
+# define PCM_CAP_ANALOGIN 0x00200000
+# define PCM_CAP_DIGITALOUT 0x00400000
+# define PCM_CAP_DIGITALIN 0x00800000
+# define PCM_CAP_ADMASK 0x00f00000
+/*
+ * NOTE! (capabilities & PCM_CAP_ADMASK)==0 means just that the
+ * digital/analog interface control features are not supported by the
+ * device/driver. However the device still supports analog, digital or
+ * both inputs/outputs (depending on the device). See the OSS Programmer's
+ * Guide for full details.
+ */
+# define PCM_CAP_SHADOW 0x01000000 /* "Shadow" device */
+
+/*
+ * Preferred channel usage. These bits can be used to
+ * give recommendations to the application. Used by few drivers.
+ * For example if ((caps & DSP_CH_MASK) == DSP_CH_MONO) means that
+ * the device works best in mono mode. However it doesn't necessarily mean
+ * that the device cannot be used in stereo. These bits should only be used
+ * by special applications such as multi track hard disk recorders to find
+ * out the initial setup. However the user should be able to override this
+ * selection.
+ *
+ * To find out which modes are actually supported the application should
+ * try to select them using SNDCTL_DSP_CHANNELS.
+ */
+# define DSP_CH_MASK 0x06000000 /* Mask */
+# define DSP_CH_ANY 0x00000000 /* No preferred mode */
+# define DSP_CH_MONO 0x02000000
+# define DSP_CH_STEREO 0x04000000
+# define DSP_CH_MULTI 0x06000000 /* More than two channels */
+
+# define PCM_CAP_HIDDEN 0x08000000 /* Hidden device */
+# define PCM_CAP_FREERATE 0x10000000
+# define PCM_CAP_MODEM 0x20000000 /* Modem device */
+# define PCM_CAP_DEFAULT 0x40000000 /* "Default" device */
+
+/*
+ * The PCM_CAP_* capability names were known as DSP_CAP_* prior OSS 4.0
+ * so it's necessary to define the older names too.
+ */
+#define DSP_CAP_ADMASK PCM_CAP_ADMASK
+#define DSP_CAP_ANALOGIN PCM_CAP_ANALOGIN
+#define DSP_CAP_ANALOGOUT PCM_CAP_ANALOGOUT
+#define DSP_CAP_BATCH PCM_CAP_BATCH
+#define DSP_CAP_BIND PCM_CAP_BIND
+#define DSP_CAP_COPROC PCM_CAP_COPROC
+#define DSP_CAP_DEFAULT PCM_CAP_DEFAULT
+#define DSP_CAP_DIGITALIN PCM_CAP_DIGITALIN
+#define DSP_CAP_DIGITALOUT PCM_CAP_DIGITALOUT
+#define DSP_CAP_DUPLEX PCM_CAP_DUPLEX
+#define DSP_CAP_FREERATE PCM_CAP_FREERATE
+#define DSP_CAP_HIDDEN PCM_CAP_HIDDEN
+#define DSP_CAP_INPUT PCM_CAP_INPUT
+#define DSP_CAP_MMAP PCM_CAP_MMAP
+#define DSP_CAP_MODEM PCM_CAP_MODEM
+#define DSP_CAP_MULTI PCM_CAP_MULTI
+#define DSP_CAP_OUTPUT PCM_CAP_OUTPUT
+#define DSP_CAP_REALTIME PCM_CAP_REALTIME
+#define DSP_CAP_REVISION PCM_CAP_REVISION
+#define DSP_CAP_SHADOW PCM_CAP_SHADOW
+#define DSP_CAP_TRIGGER PCM_CAP_TRIGGER
+#define DSP_CAP_VIRTUAL PCM_CAP_VIRTUAL
+
+#define SNDCTL_DSP_GETTRIGGER __SIOR ('P',16, int)
+#define SNDCTL_DSP_SETTRIGGER __SIOW ('P',16, int)
+# define PCM_ENABLE_INPUT 0x00000001
+# define PCM_ENABLE_OUTPUT 0x00000002
+
+typedef struct count_info
+{
+ unsigned int bytes; /* Total # of bytes processed */
+ int blocks; /* # of fragment transitions since last time */
+ int ptr; /* Current DMA pointer value */
+} count_info;
+
+#define SNDCTL_DSP_GETIPTR __SIOR ('P',17, count_info)
+#define SNDCTL_DSP_GETOPTR __SIOR ('P',18, count_info)
+
+typedef struct buffmem_desc
+{
+ unsigned *buffer;
+ int size;
+} buffmem_desc;
+#define SNDCTL_DSP_SETSYNCRO __SIO ('P', 21)
+#define SNDCTL_DSP_SETDUPLEX __SIO ('P', 22)
+
+#define SNDCTL_DSP_PROFILE __SIOW ('P', 23, int) /* OBSOLETE */
+#define APF_NORMAL 0 /* Normal applications */
+#define APF_NETWORK 1 /* Underruns probably caused by an "external" delay */
+#define APF_CPUINTENS 2 /* Underruns probably caused by "overheating" the CPU */
+
+#define SNDCTL_DSP_GETODELAY __SIOR ('P', 23, int)
+
+typedef struct audio_errinfo
+{
+ int play_underruns;
+ int rec_overruns;
+ unsigned int play_ptradjust;
+ unsigned int rec_ptradjust;
+ int play_errorcount;
+ int rec_errorcount;
+ int play_lasterror;
+ int rec_lasterror;
+ int play_errorparm;
+ int rec_errorparm;
+ int filler[16];
+} audio_errinfo;
+
+#define SNDCTL_DSP_GETPLAYVOL __SIOR ('P', 24, int)
+#define SNDCTL_DSP_SETPLAYVOL __SIOWR('P', 24, int)
+#define SNDCTL_DSP_GETERROR __SIOR ('P', 25, audio_errinfo)
+/*
+ ****************************************************************************
+ * Digital interface (S/PDIF) control interface
+ */
+
+typedef struct oss_digital_control
+{
+ unsigned int caps;
+#define DIG_CBITIN_NONE 0x00000000
+#define DIG_CBITIN_LIMITED 0x00000001
+#define DIG_CBITIN_DATA 0x00000002
+#define DIG_CBITIN_BYTE0 0x00000004
+#define DIG_CBITIN_FULL 0x00000008
+#define DIG_CBITIN_MASK 0x0000000f
+#define DIG_CBITOUT_NONE 0x00000000
+#define DIG_CBITOUT_LIMITED 0x00000010
+#define DIG_CBITOUT_BYTE0 0x00000020
+#define DIG_CBITOUT_FULL 0x00000040
+#define DIG_CBITOUT_DATA 0x00000080
+#define DIG_CBITOUT_MASK 0x000000f0
+#define DIG_UBITIN 0x00000100
+#define DIG_UBITOUT 0x00000200
+#define DIG_VBITOUT 0x00000400
+#define DIG_OUTRATE 0x00000800
+#define DIG_INRATE 0x00001000
+#define DIG_INBITS 0x00002000
+#define DIG_OUTBITS 0x00004000
+#define DIG_EXACT 0x00010000
+#define DIG_PRO 0x00020000
+#define DIG_CONSUMER 0x00040000
+#define DIG_PASSTHROUGH 0x00080000
+#define DIG_OUTSEL 0x00100000
+
+ unsigned int valid;
+#define VAL_CBITIN 0x00000001
+#define VAL_UBITIN 0x00000002
+#define VAL_CBITOUT 0x00000004
+#define VAL_UBITOUT 0x00000008
+#define VAL_ISTATUS 0x00000010
+#define VAL_IRATE 0x00000020
+#define VAL_ORATE 0x00000040
+#define VAL_INBITS 0x00000080
+#define VAL_OUTBITS 0x00000100
+#define VAL_REQUEST 0x00000200
+#define VAL_OUTSEL 0x00000400
+
+#define VAL_OUTMASK (VAL_CBITOUT|VAL_UBITOUT|VAL_ORATE|VAL_OUTBITS|VAL_OUTSEL)
+
+ unsigned int request, param;
+#define SPD_RQ_PASSTHROUGH 1
+
+ unsigned char cbitin[24];
+ unsigned char ubitin[24];
+ unsigned char cbitout[24];
+ unsigned char ubitout[24];
+
+ unsigned int outsel;
+#define OUTSEL_DIGITAL 1
+#define OUTSEL_ANALOG 2
+#define OUTSEL_BOTH (OUTSEL_DIGITAL|OUTSEL_ANALOG)
+
+ int in_data; /* Audio/data if autodetectable by the receiver */
+#define IND_UNKNOWN 0
+#define IND_AUDIO 1
+#define IND_DATA 2
+
+ int in_locked; /* Receiver locked */
+#define LOCK_NOT_INDICATED 0
+#define LOCK_UNLOCKED 1
+#define LOCK_LOCKED 2
+
+ int in_quality; /* Input signal quality */
+#define IN_QUAL_NOT_INDICATED 0
+#define IN_QUAL_POOR 1
+#define IN_QUAL_GOOD 2
+
+ int in_vbit, out_vbit; /* V bits */
+#define VBIT_NOT_INDICATED 0
+#define VBIT_OFF 1
+#define VBIT_ON 2
+
+ unsigned int in_errors; /* Various input error conditions */
+#define INERR_CRC 0x0001
+#define INERR_QCODE_CRC 0x0002
+#define INERR_PARITY 0x0004
+#define INERR_BIPHASE 0x0008
+
+ int srate_in, srate_out;
+ int bits_in, bits_out;
+
+ int filler[32];
+} oss_digital_control;
+
+#define SNDCTL_DSP_READCTL __SIOWR('P', 26, oss_digital_control)
+#define SNDCTL_DSP_WRITECTL __SIOWR('P', 27, oss_digital_control)
+
+/*
+ ****************************************************************************
+ * Sync groups for audio devices
+ */
+typedef struct oss_syncgroup
+{
+ int id;
+ int mode;
+ int filler[16];
+} oss_syncgroup;
+
+#define SNDCTL_DSP_SYNCGROUP __SIOWR('P', 28, oss_syncgroup)
+#define SNDCTL_DSP_SYNCSTART __SIOW ('P', 29, int)
+
+/*
+ **************************************************************************
+ * "cooked" mode enables software based conversions for sample rate, sample
+ * format (bits) and number of channels (mono/stereo). These conversions are
+ * required with some devices that support only one sample rate or just stereo
+ * to let the applications to use other formats. The cooked mode is enabled by
+ * default. However it's necessary to disable this mode when mmap() is used or
+ * when very deterministic timing is required. SNDCTL_DSP_COOKEDMODE is an
+ * optional call introduced in OSS 3.9.6f. It's _error return must be ignored_
+ * since normally this call will return erno=EINVAL.
+ *
+ * SNDCTL_DSP_COOKEDMODE must be called immediately after open before doing
+ * anything else. Otherwise the call will not have any effect.
+ */
+#define SNDCTL_DSP_COOKEDMODE __SIOW ('P', 30, int)
+
+/*
+ **************************************************************************
+ * SNDCTL_DSP_SILENCE and SNDCTL_DSP_SKIP are new calls in OSS 3.99.0
+ * that can be used to implement pause/continue during playback (no effect
+ * on recording).
+ */
+#define SNDCTL_DSP_SILENCE __SIO ('P', 31)
+#define SNDCTL_DSP_SKIP __SIO ('P', 32)
+/*
+ ****************************************************************************
+ * Abort transfer (reset) functions for input and output
+ */
+#define SNDCTL_DSP_HALT_INPUT __SIO ('P', 33)
+#define SNDCTL_DSP_RESET_INPUT SNDCTL_DSP_HALT_INPUT /* Old name */
+#define SNDCTL_DSP_HALT_OUTPUT __SIO ('P', 34)
+#define SNDCTL_DSP_RESET_OUTPUT SNDCTL_DSP_HALT_OUTPUT /* Old name */
+/*
+ ****************************************************************************
+ * Low water level control
+ */
+#define SNDCTL_DSP_LOW_WATER __SIOW ('P', 34, int)
+
+/*
+ ****************************************************************************
+ * 64 bit pointer support. Only available in environments that support
+ * the 64 bit (long long) integer type.
+ */
+#ifndef OSS_NO_LONG_LONG
+typedef struct
+{
+ long long samples;
+ int fifo_samples;
+ int filler[32]; /* For future use */
+} oss_count_t;
+
+#define SNDCTL_DSP_CURRENT_IPTR __SIOR ('P', 35, oss_count_t)
+#define SNDCTL_DSP_CURRENT_OPTR __SIOR ('P', 36, oss_count_t)
+#endif
+
+/*
+ ****************************************************************************
+ * Interface for selecting recording sources and playback output routings.
+ */
+#define SNDCTL_DSP_GET_RECSRC_NAMES __SIOR ('P', 37, oss_mixer_enuminfo)
+#define SNDCTL_DSP_GET_RECSRC __SIOR ('P', 38, int)
+#define SNDCTL_DSP_SET_RECSRC __SIOWR('P', 38, int)
+
+#define SNDCTL_DSP_GET_PLAYTGT_NAMES __SIOR ('P', 39, oss_mixer_enuminfo)
+#define SNDCTL_DSP_GET_PLAYTGT __SIOR ('P', 40, int)
+#define SNDCTL_DSP_SET_PLAYTGT __SIOWR('P', 40, int)
+#define SNDCTL_DSP_GETRECVOL __SIOR ('P', 41, int)
+#define SNDCTL_DSP_SETRECVOL __SIOWR('P', 41, int)
+
+/*
+ ***************************************************************************
+ * Some calls for setting the channel assignment with multi channel devices
+ * (see the manual for details).
+ */
+#ifndef OSS_NO_LONG_LONG
+#define SNDCTL_DSP_GET_CHNORDER __SIOR ('P', 42, unsigned long long)
+#define SNDCTL_DSP_SET_CHNORDER __SIOWR('P', 42, unsigned long long)
+# define CHID_UNDEF 0
+# define CHID_L 1
+# define CHID_R 2
+# define CHID_C 3
+# define CHID_LFE 4
+# define CHID_LS 5
+# define CHID_RS 6
+# define CHID_LR 7
+# define CHID_RR 8
+#define CHNORDER_UNDEF 0x0000000000000000ULL
+#define CHNORDER_NORMAL 0x0000000087654321ULL
+#endif
+
+#define MAX_PEAK_CHANNELS 128
+typedef unsigned short oss_peaks_t[MAX_PEAK_CHANNELS];
+#define SNDCTL_DSP_GETIPEAKS __SIOR('P', 43, oss_peaks_t)
+#define SNDCTL_DSP_GETOPEAKS __SIOR('P', 44, oss_peaks_t)
+
+#define SNDCTL_DSP_POLICY __SIOW('P', 45, int) /* See the manual */
+
+/*
+ ****************************************************************************
+ * Few ioctl calls that are not official parts of OSS. They have been used
+ * by few freeware implementations of OSS.
+ */
+#define SNDCTL_DSP_GETCHANNELMASK __SIOWR('P', 64, int)
+#define SNDCTL_DSP_BIND_CHANNEL __SIOWR('P', 65, int)
+# define DSP_BIND_QUERY 0x00000000
+# define DSP_BIND_FRONT 0x00000001
+# define DSP_BIND_SURR 0x00000002
+# define DSP_BIND_CENTER_LFE 0x00000004
+# define DSP_BIND_HANDSET 0x00000008
+# define DSP_BIND_MIC 0x00000010
+# define DSP_BIND_MODEM1 0x00000020
+# define DSP_BIND_MODEM2 0x00000040
+# define DSP_BIND_I2S 0x00000080
+# define DSP_BIND_SPDIF 0x00000100
+# define DSP_BIND_REAR 0x00000200
+
+#ifdef sun
+/* Not part of OSS. Reserved for internal use by Solaris */
+#define X_SADA_GET_PLAYTGT_MASK __SIOR ('P', 66, int)
+#define X_SADA_GET_PLAYTGT __SIOR ('P', 67, int)
+#define X_SADA_SET_PLAYTGT __SIOWR('P', 68, int)
+#endif
+
+#ifndef NO_LEGACY_MIXER
+/*
+ ****************************************************************************
+ * IOCTL commands for the "legacy " /dev/mixer API (obsolete)
+ *
+ * Mixer controls
+ *
+ * There can be up to 20 different analog mixer channels. The
+ * SOUND_MIXER_NRDEVICES gives the currently supported maximum.
+ * The SOUND_MIXER_READ_DEVMASK returns a bitmask which tells
+ * the devices supported by the particular mixer.
+ *
+ * {!notice This "legacy" mixer API is obsolete. It has been superceded
+ * by a new one (see below).
+ */
+
+#define SOUND_MIXER_NRDEVICES 28
+#define SOUND_MIXER_VOLUME 0
+#define SOUND_MIXER_BASS 1
+#define SOUND_MIXER_TREBLE 2
+#define SOUND_MIXER_SYNTH 3
+#define SOUND_MIXER_PCM 4
+#define SOUND_MIXER_SPEAKER 5
+#define SOUND_MIXER_LINE 6
+#define SOUND_MIXER_MIC 7
+#define SOUND_MIXER_CD 8
+#define SOUND_MIXER_IMIX 9 /* Recording monitor */
+#define SOUND_MIXER_ALTPCM 10
+#define SOUND_MIXER_RECLEV 11 /* Recording level */
+#define SOUND_MIXER_IGAIN 12 /* Input gain */
+#define SOUND_MIXER_OGAIN 13 /* Output gain */
+/*
+ * Some soundcards have three line level inputs (line, aux1 and aux2).
+ * Since each card manufacturer has assigned different meanings to
+ * these inputs, it's impractical to assign specific meanings
+ * (eg line, cd, synth etc.) to them.
+ */
+#define SOUND_MIXER_LINE1 14 /* Input source 1 (aux1) */
+#define SOUND_MIXER_LINE2 15 /* Input source 2 (aux2) */
+#define SOUND_MIXER_LINE3 16 /* Input source 3 (line) */
+#define SOUND_MIXER_DIGITAL1 17 /* Digital I/O 1 */
+#define SOUND_MIXER_DIGITAL2 18 /* Digital I/O 2 */
+#define SOUND_MIXER_DIGITAL3 19 /* Digital I/O 3 */
+#define SOUND_MIXER_PHONE 20 /* Phone */
+#define SOUND_MIXER_MONO 21 /* Mono Output */
+#define SOUND_MIXER_VIDEO 22 /* Video/TV (audio) in */
+#define SOUND_MIXER_RADIO 23 /* Radio in */
+#define SOUND_MIXER_DEPTH 24 /* Surround depth */
+#define SOUND_MIXER_REARVOL 25 /* Rear/Surround speaker vol */
+#define SOUND_MIXER_CENTERVOL 26 /* Center/LFE speaker vol */
+#define SOUND_MIXER_SIDEVOL 27 /* Side-Surround (8speaker) vol */
+
+/*
+ * Warning: SOUND_MIXER_SURRVOL is an old name of SOUND_MIXER_SIDEVOL.
+ * They are both assigned to the same mixer control. Don't
+ * use both control names in the same program/driver.
+ */
+#define SOUND_MIXER_SURRVOL SOUND_MIXER_SIDEVOL
+
+/* Some on/off settings (SOUND_SPECIAL_MIN - SOUND_SPECIAL_MAX) */
+/* Not counted to SOUND_MIXER_NRDEVICES, but use the same number space */
+#define SOUND_ONOFF_MIN 28
+#define SOUND_ONOFF_MAX 30
+
+/* Note! Number 31 cannot be used since the sign bit is reserved */
+#define SOUND_MIXER_NONE 31
+
+/*
+ * The following unsupported macros are no longer functional.
+ * Use SOUND_MIXER_PRIVATE# macros in future.
+ */
+#define SOUND_MIXER_ENHANCE SOUND_MIXER_NONE
+#define SOUND_MIXER_MUTE SOUND_MIXER_NONE
+#define SOUND_MIXER_LOUD SOUND_MIXER_NONE
+
+#define SOUND_DEVICE_LABELS \
+ {"Vol ", "Bass ", "Treble", "Synth", "Pcm ", "Speaker ", "Line ", \
+ "Mic ", "CD ", "Mix ", "Pcm2 ", "Rec ", "IGain", "OGain", \
+ "Aux1", "Aux2", "Aux3", "Digital1", "Digital2", "Digital3", \
+ "Phone", "Mono", "Video", "Radio", "Depth", \
+ "Rear", "Center", "Side"}
+
+#define SOUND_DEVICE_NAMES \
+ {"vol", "bass", "treble", "synth", "pcm", "speaker", "line", \
+ "mic", "cd", "mix", "pcm2", "rec", "igain", "ogain", \
+ "aux1", "aux2", "aux3", "dig1", "dig2", "dig3", \
+ "phone", "mono", "video", "radio", "depth", \
+ "rear", "center", "side"}
+
+/* Device bitmask identifiers */
+
+#define SOUND_MIXER_RECSRC 0xff /* Arg contains a bit for each recording source */
+#define SOUND_MIXER_DEVMASK 0xfe /* Arg contains a bit for each supported device */
+#define SOUND_MIXER_RECMASK 0xfd /* Arg contains a bit for each supported recording source */
+#define SOUND_MIXER_CAPS 0xfc
+# define SOUND_CAP_EXCL_INPUT 0x00000001 /* Only one recording source at a time */
+# define SOUND_CAP_NOLEGACY 0x00000004 /* For internal use only */
+# define SOUND_CAP_NORECSRC 0x00000008
+#define SOUND_MIXER_STEREODEVS 0xfb /* Mixer channels supporting stereo */
+
+/* OSS/Free ONLY */
+#define SOUND_MIXER_OUTSRC 0xfa /* Arg contains a bit for each input source to output */
+#define SOUND_MIXER_OUTMASK 0xf9 /* Arg contains a bit for each supported input source to output */
+/* OSS/Free ONLY */
+
+/* Device mask bits */
+
+#define SOUND_MASK_VOLUME (1 << SOUND_MIXER_VOLUME)
+#define SOUND_MASK_BASS (1 << SOUND_MIXER_BASS)
+#define SOUND_MASK_TREBLE (1 << SOUND_MIXER_TREBLE)
+#define SOUND_MASK_SYNTH (1 << SOUND_MIXER_SYNTH)
+#define SOUND_MASK_PCM (1 << SOUND_MIXER_PCM)
+#define SOUND_MASK_SPEAKER (1 << SOUND_MIXER_SPEAKER)
+#define SOUND_MASK_LINE (1 << SOUND_MIXER_LINE)
+#define SOUND_MASK_MIC (1 << SOUND_MIXER_MIC)
+#define SOUND_MASK_CD (1 << SOUND_MIXER_CD)
+#define SOUND_MASK_IMIX (1 << SOUND_MIXER_IMIX)
+#define SOUND_MASK_ALTPCM (1 << SOUND_MIXER_ALTPCM)
+#define SOUND_MASK_RECLEV (1 << SOUND_MIXER_RECLEV)
+#define SOUND_MASK_IGAIN (1 << SOUND_MIXER_IGAIN)
+#define SOUND_MASK_OGAIN (1 << SOUND_MIXER_OGAIN)
+#define SOUND_MASK_LINE1 (1 << SOUND_MIXER_LINE1)
+#define SOUND_MASK_LINE2 (1 << SOUND_MIXER_LINE2)
+#define SOUND_MASK_LINE3 (1 << SOUND_MIXER_LINE3)
+#define SOUND_MASK_DIGITAL1 (1 << SOUND_MIXER_DIGITAL1)
+#define SOUND_MASK_DIGITAL2 (1 << SOUND_MIXER_DIGITAL2)
+#define SOUND_MASK_DIGITAL3 (1 << SOUND_MIXER_DIGITAL3)
+#define SOUND_MASK_MONO (1 << SOUND_MIXER_MONO)
+#define SOUND_MASK_PHONE (1 << SOUND_MIXER_PHONE)
+#define SOUND_MASK_RADIO (1 << SOUND_MIXER_RADIO)
+#define SOUND_MASK_VIDEO (1 << SOUND_MIXER_VIDEO)
+#define SOUND_MASK_DEPTH (1 << SOUND_MIXER_DEPTH)
+#define SOUND_MASK_REARVOL (1 << SOUND_MIXER_REARVOL)
+#define SOUND_MASK_CENTERVOL (1 << SOUND_MIXER_CENTERVOL)
+#define SOUND_MASK_SIDEVOL (1 << SOUND_MIXER_SIDEVOL)
+
+/* Note! SOUND_MASK_SURRVOL is alias of SOUND_MASK_SIDEVOL */
+#define SOUND_MASK_SURRVOL (1 << SOUND_MIXER_SIDEVOL)
+
+/* Obsolete macros */
+#define SOUND_MASK_MUTE (1 << SOUND_MIXER_MUTE)
+#define SOUND_MASK_ENHANCE (1 << SOUND_MIXER_ENHANCE)
+#define SOUND_MASK_LOUD (1 << SOUND_MIXER_LOUD)
+
+#define MIXER_READ(dev) __SIOR('M', dev, int)
+#define SOUND_MIXER_READ_VOLUME MIXER_READ(SOUND_MIXER_VOLUME)
+#define SOUND_MIXER_READ_BASS MIXER_READ(SOUND_MIXER_BASS)
+#define SOUND_MIXER_READ_TREBLE MIXER_READ(SOUND_MIXER_TREBLE)
+#define SOUND_MIXER_READ_SYNTH MIXER_READ(SOUND_MIXER_SYNTH)
+#define SOUND_MIXER_READ_PCM MIXER_READ(SOUND_MIXER_PCM)
+#define SOUND_MIXER_READ_SPEAKER MIXER_READ(SOUND_MIXER_SPEAKER)
+#define SOUND_MIXER_READ_LINE MIXER_READ(SOUND_MIXER_LINE)
+#define SOUND_MIXER_READ_MIC MIXER_READ(SOUND_MIXER_MIC)
+#define SOUND_MIXER_READ_CD MIXER_READ(SOUND_MIXER_CD)
+#define SOUND_MIXER_READ_IMIX MIXER_READ(SOUND_MIXER_IMIX)
+#define SOUND_MIXER_READ_ALTPCM MIXER_READ(SOUND_MIXER_ALTPCM)
+#define SOUND_MIXER_READ_RECLEV MIXER_READ(SOUND_MIXER_RECLEV)
+#define SOUND_MIXER_READ_IGAIN MIXER_READ(SOUND_MIXER_IGAIN)
+#define SOUND_MIXER_READ_OGAIN MIXER_READ(SOUND_MIXER_OGAIN)
+#define SOUND_MIXER_READ_LINE1 MIXER_READ(SOUND_MIXER_LINE1)
+#define SOUND_MIXER_READ_LINE2 MIXER_READ(SOUND_MIXER_LINE2)
+#define SOUND_MIXER_READ_LINE3 MIXER_READ(SOUND_MIXER_LINE3)
+
+/* Obsolete macros */
+#define SOUND_MIXER_READ_MUTE MIXER_READ(SOUND_MIXER_MUTE)
+#define SOUND_MIXER_READ_ENHANCE MIXER_READ(SOUND_MIXER_ENHANCE)
+#define SOUND_MIXER_READ_LOUD MIXER_READ(SOUND_MIXER_LOUD)
+
+#define SOUND_MIXER_READ_RECSRC MIXER_READ(SOUND_MIXER_RECSRC)
+#define SOUND_MIXER_READ_DEVMASK MIXER_READ(SOUND_MIXER_DEVMASK)
+#define SOUND_MIXER_READ_RECMASK MIXER_READ(SOUND_MIXER_RECMASK)
+#define SOUND_MIXER_READ_STEREODEVS MIXER_READ(SOUND_MIXER_STEREODEVS)
+#define SOUND_MIXER_READ_CAPS MIXER_READ(SOUND_MIXER_CAPS)
+
+#define MIXER_WRITE(dev) __SIOWR('M', dev, int)
+#define SOUND_MIXER_WRITE_VOLUME MIXER_WRITE(SOUND_MIXER_VOLUME)
+#define SOUND_MIXER_WRITE_BASS MIXER_WRITE(SOUND_MIXER_BASS)
+#define SOUND_MIXER_WRITE_TREBLE MIXER_WRITE(SOUND_MIXER_TREBLE)
+#define SOUND_MIXER_WRITE_SYNTH MIXER_WRITE(SOUND_MIXER_SYNTH)
+#define SOUND_MIXER_WRITE_PCM MIXER_WRITE(SOUND_MIXER_PCM)
+#define SOUND_MIXER_WRITE_SPEAKER MIXER_WRITE(SOUND_MIXER_SPEAKER)
+#define SOUND_MIXER_WRITE_LINE MIXER_WRITE(SOUND_MIXER_LINE)
+#define SOUND_MIXER_WRITE_MIC MIXER_WRITE(SOUND_MIXER_MIC)
+#define SOUND_MIXER_WRITE_CD MIXER_WRITE(SOUND_MIXER_CD)
+#define SOUND_MIXER_WRITE_IMIX MIXER_WRITE(SOUND_MIXER_IMIX)
+#define SOUND_MIXER_WRITE_ALTPCM MIXER_WRITE(SOUND_MIXER_ALTPCM)
+#define SOUND_MIXER_WRITE_RECLEV MIXER_WRITE(SOUND_MIXER_RECLEV)
+#define SOUND_MIXER_WRITE_IGAIN MIXER_WRITE(SOUND_MIXER_IGAIN)
+#define SOUND_MIXER_WRITE_OGAIN MIXER_WRITE(SOUND_MIXER_OGAIN)
+#define SOUND_MIXER_WRITE_LINE1 MIXER_WRITE(SOUND_MIXER_LINE1)
+#define SOUND_MIXER_WRITE_LINE2 MIXER_WRITE(SOUND_MIXER_LINE2)
+#define SOUND_MIXER_WRITE_LINE3 MIXER_WRITE(SOUND_MIXER_LINE3)
+
+/* Obsolete macros */
+#define SOUND_MIXER_WRITE_MUTE MIXER_WRITE(SOUND_MIXER_MUTE)
+#define SOUND_MIXER_WRITE_ENHANCE MIXER_WRITE(SOUND_MIXER_ENHANCE)
+#define SOUND_MIXER_WRITE_LOUD MIXER_WRITE(SOUND_MIXER_LOUD)
+
+#define SOUND_MIXER_WRITE_RECSRC MIXER_WRITE(SOUND_MIXER_RECSRC)
+
+typedef struct mixer_info /* OBSOLETE */
+{
+ char id[16];
+ char name[32];
+ int modify_counter;
+ int card_number;
+ int port_number;
+ char handle[32];
+} mixer_info;
+
+/* SOUND_MIXER_INFO is obsolete - use SNDCTL_MIXERINFO instead */
+#define SOUND_MIXER_INFO __SIOR ('M', 101, mixer_info)
+
+/*
+ * Two ioctls for special souncard function (OSS/Free only)
+ */
+#define SOUND_MIXER_AGC _SIOWR('M', 103, int)
+#define SOUND_MIXER_3DSE _SIOWR('M', 104, int)
+/*
+ * The SOUND_MIXER_PRIVATE# commands can be redefined by low level drivers.
+ * These features can be used when accessing device specific features.
+ */
+#define SOUND_MIXER_PRIVATE1 __SIOWR('M', 111, int)
+#define SOUND_MIXER_PRIVATE2 __SIOWR('M', 112, int)
+#define SOUND_MIXER_PRIVATE3 __SIOWR('M', 113, int)
+#define SOUND_MIXER_PRIVATE4 __SIOWR('M', 114, int)
+#define SOUND_MIXER_PRIVATE5 __SIOWR('M', 115, int)
+
+/* The following two controls were never implemented and they should not be used. */
+#define SOUND_MIXER_READ_MAINVOL __SIOR ('M', 116, int)
+#define SOUND_MIXER_WRITE_MAINVOL __SIOWR('M', 116, int)
+
+/*
+ * SOUND_MIXER_GETLEVELS and SOUND_MIXER_SETLEVELS calls can be used
+ * for querying current mixer settings from the driver and for loading
+ * default volume settings _prior_ activating the mixer (loading
+ * doesn't affect current state of the mixer hardware). These calls
+ * are for internal use by the driver software only.
+ */
+
+typedef struct mixer_vol_table
+{
+ int num; /* Index to volume table */
+ char name[32];
+ int levels[32];
+} mixer_vol_table;
+
+#define SOUND_MIXER_GETLEVELS __SIOWR('M', 116, mixer_vol_table)
+#define SOUND_MIXER_SETLEVELS __SIOWR('M', 117, mixer_vol_table)
+
+#define OSS_GETVERSION __SIOR ('M', 118, int)
+
+/*
+ * Calls to set/get the recording gain for the currently active
+ * recording source. These calls automatically map to the right control.
+ * Note that these calls are not supported by all drivers. In this case
+ * the call will return -1 with errno set to EINVAL
+ *
+ * The _MONGAIN work in similar way but set/get the monitoring gain for
+ * the currently selected recording source.
+ */
+#define SOUND_MIXER_READ_RECGAIN __SIOR ('M', 119, int)
+#define SOUND_MIXER_WRITE_RECGAIN __SIOWR('M', 119, int)
+#define SOUND_MIXER_READ_MONGAIN __SIOR ('M', 120, int)
+#define SOUND_MIXER_WRITE_MONGAIN __SIOWR('M', 120, int)
+
+/* The following call is for driver development time purposes. It's not
+ * present in any released drivers.
+ */
+typedef unsigned char oss_reserved_t[512];
+#define SOUND_MIXER_RESERVED __SIOWR('M', 121, oss_reserved_t)
+#endif /* ifndef NO_LEGACY_MIXER */
+
+/*
+ *************************************************************************
+ * The "new" mixer API of OSS 4.0 and later.
+ *
+ * This improved mixer API makes it possible to access every possible feature
+ * of every possible device. However you should read the mixer programming
+ * section of the OSS API Developer's Manual. There is no chance that you
+ * could use this interface correctly just by examining this header.
+ */
+
+typedef struct oss_sysinfo
+{
+ char product[32]; /* For example OSS/Free, OSS/Linux or OSS/Solaris */
+ char version[32]; /* For example 4.0a */
+ int versionnum; /* See OSS_GETVERSION */
+ char options[128]; /* Reserved */
+
+ int numaudios; /* # of audio/dsp devices */
+ int openedaudio[8]; /* Bit mask telling which audio devices are busy */
+
+ int numsynths; /* # of availavle synth devices */
+ int nummidis; /* # of available MIDI ports */
+ int numtimers; /* # of available timer devices */
+ int nummixers; /* # of mixer devices */
+
+ int openedmidi[8]; /* Bit mask telling which midi devices are busy */
+ int numcards; /* Number of sound cards in the system */
+ int numaudioengines; /* Number of audio engines in the system */
+ char license[16]; /* For example "GPL" or "CDDL" */
+ int filler[236]; /* For future expansion (set to -1) */
+} oss_sysinfo;
+
+typedef struct oss_mixext
+{
+ int dev; /* Mixer device number */
+ int ctrl; /* Controller number */
+ int type; /* Entry type */
+# define MIXT_DEVROOT 0 /* Device root entry */
+# define MIXT_GROUP 1 /* Controller group */
+# define MIXT_ONOFF 2 /* OFF (0) or ON (1) */
+# define MIXT_ENUM 3 /* Enumerated (0 to maxvalue) */
+# define MIXT_MONOSLIDER 4 /* Mono slider (0 to 255) */
+# define MIXT_STEREOSLIDER 5 /* Stereo slider (dual 0 to 255) */
+# define MIXT_MESSAGE 6 /* (Readable) textual message */
+# define MIXT_MONOVU 7 /* VU meter value (mono) */
+# define MIXT_STEREOVU 8 /* VU meter value (stereo) */
+# define MIXT_MONOPEAK 9 /* VU meter peak value (mono) */
+# define MIXT_STEREOPEAK 10 /* VU meter peak value (stereo) */
+# define MIXT_RADIOGROUP 11 /* Radio button group */
+# define MIXT_MARKER 12 /* Separator between normal and extension entries */
+# define MIXT_VALUE 13 /* Decimal value entry */
+# define MIXT_HEXVALUE 14 /* Hexadecimal value entry */
+# define MIXT_MONODB 15 /* OBSOLETE */
+# define MIXT_STEREODB 16 /* OBSOLETE */
+# define MIXT_SLIDER 17 /* Slider (mono) with full (31 bit) postitive integer range */
+# define MIXT_3D 18
+
+/*
+ * Sliders with range expanded to 15 bits per channel (0-32767)
+ */
+# define MIXT_MONOSLIDER16 19
+# define MIXT_STEREOSLIDER16 20
+# define MIXT_MUTE 21 /* Mute=1, unmute=0 */
+
+ /**************************************************************/
+
+ /* Possible value range (minvalue to maxvalue) */
+ /* Note that maxvalue may also be smaller than minvalue */
+ int maxvalue;
+ int minvalue;
+
+ int flags;
+# define MIXF_READABLE 0x00000001 /* Has readable value */
+# define MIXF_WRITEABLE 0x00000002 /* Has writeable value */
+# define MIXF_POLL 0x00000004 /* May change itself */
+# define MIXF_HZ 0x00000008 /* Herz scale */
+# define MIXF_STRING 0x00000010 /* Use dynamic extensions for value */
+# define MIXF_DYNAMIC 0x00000010 /* Supports dynamic extensions */
+# define MIXF_OKFAIL 0x00000020 /* Interpret value as 1=OK, 0=FAIL */
+# define MIXF_FLAT 0x00000040 /* Flat vertical space requirements */
+# define MIXF_LEGACY 0x00000080 /* Legacy mixer control group */
+# define MIXF_CENTIBEL 0x00000100 /* Centibel (0.1 dB) step size */
+# define MIXF_DECIBEL 0x00000200 /* Step size of 1 dB */
+# define MIXF_MAINVOL 0x00000400 /* Main volume control */
+# define MIXF_PCMVOL 0x00000800 /* PCM output volume control */
+# define MIXF_RECVOL 0x00001000 /* PCM recording volume control */
+# define MIXF_MONVOL 0x00002000 /* Input->output monitor volume */
+# define MIXF_WIDE 0x00004000 /* Enum control has wide labels */
+# define MIXF_DESCR 0x00008000 /* Description (tooltip) available */
+ char id[16]; /* Mnemonic ID (mainly for internal use) */
+ int parent; /* Entry# of parent (group) node (-1 if root) */
+
+ int dummy; /* Internal use */
+
+ int timestamp;
+
+ char data[64]; /* Misc data (entry type dependent) */
+ unsigned char enum_present[32]; /* Mask of allowed enum values */
+ int control_no; /* SOUND_MIXER_VOLUME..SOUND_MIXER_MIDI */
+ /* (-1 means not indicated) */
+
+/*
+ * The desc field is reserved for internal purposes of OSS. It should not be
+ * used by applications.
+ */
+ unsigned int desc;
+#define MIXEXT_SCOPE_MASK 0x0000003f
+#define MIXEXT_SCOPE_OTHER 0x00000000
+#define MIXEXT_SCOPE_INPUT 0x00000001
+#define MIXEXT_SCOPE_OUTPUT 0x00000002
+#define MIXEXT_SCOPE_MONITOR 0x00000003
+#define MIXEXT_SCOPE_RECSWITCH 0x00000004
+
+ char extname[32];
+ int update_counter;
+ int rgbcolor; /* 0 means default color (not black) . Otherwise 24 bit RGB color */
+ int filler[6];
+} oss_mixext;
+
+/*
+ * Recommended colors to be used in the rgbcolor field. These match the
+ * colors used as the audio jack colors in HD audio motherboards.
+ */
+#define OSS_RGB_BLUE 0x7aabde // Light blue
+#define OSS_RGB_GREEN 0xb3c98c // Lime green
+#define OSS_RGB_PINK 0xe88c99
+#define OSS_RGB_GRAY 0xd1ccc4
+#define OSS_RGB_BLACK 0x2b2926 // Light black
+#define OSS_RGB_ORANGE 0xe89e47
+#define OSS_RGB_RED 0xff0000
+#define OSS_RGB_YELLOW 0xffff00
+#define OSS_RGB_PURPLE 0x800080
+#define OSS_RGB_WHITE 0xf8f8ff
+
+typedef struct oss_mixext_root
+{
+ char id[16];
+ char name[48];
+} oss_mixext_root;
+
+typedef struct oss_mixer_value
+{
+ int dev;
+ int ctrl;
+ int value;
+ int flags; /* Reserved for future use. Initialize to 0 */
+ int timestamp; /* Must be set to oss_mixext.timestamp */
+ int filler[8]; /* Reserved for future use. Initialize to 0 */
+} oss_mixer_value;
+
+#define OSS_ENUM_MAXVALUE 255
+#define OSS_ENUM_STRINGSIZE 3000
+typedef struct oss_mixer_enuminfo
+{
+ int dev;
+ int ctrl;
+ int nvalues;
+ int version; /* Read the manual */
+ short strindex[OSS_ENUM_MAXVALUE];
+ char strings[OSS_ENUM_STRINGSIZE];
+} oss_mixer_enuminfo;
+
+#define OPEN_READ PCM_ENABLE_INPUT
+#define OPEN_WRITE PCM_ENABLE_OUTPUT
+#define OPEN_READWRITE (OPEN_READ|OPEN_WRITE)
+
+typedef struct oss_audioinfo
+{
+ int dev; /* Audio device number */
+ char name[64];
+ int busy; /* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */
+ int pid;
+ int caps; /* PCM_CAP_INPUT, PCM_CAP_OUTPUT */
+ int iformats, oformats;
+ int magic; /* Reserved for internal use */
+ char cmd[64]; /* Command using the device (if known) */
+ int card_number;
+ int port_number;
+ int mixer_dev;
+ int legacy_device; /* Obsolete field. Replaced by devnode */
+ int enabled; /* 1=enabled, 0=device not ready at this moment */
+ int flags; /* For internal use only - no practical meaning */
+ int min_rate, max_rate; /* Sample rate limits */
+ int min_channels, max_channels; /* Number of channels supported */
+ int binding; /* DSP_BIND_FRONT, etc. 0 means undefined */
+ int rate_source;
+ char handle[32];
+#define OSS_MAX_SAMPLE_RATES 20 /* Cannot be changed */
+ unsigned int nrates, rates[OSS_MAX_SAMPLE_RATES]; /* Please read the manual before using these */
+ oss_longname_t song_name; /* Song name (if given) */
+ oss_label_t label; /* Device label (if given) */
+ int latency; /* In usecs, -1=unknown */
+ oss_devnode_t devnode; /* Device special file name (absolute path) */
+ int next_play_engine; /* Read the documentation for more info */
+ int next_rec_engine; /* Read the documentation for more info */
+ int filler[184];
+} oss_audioinfo;
+
+typedef struct oss_mixerinfo
+{
+ int dev;
+ char id[16];
+ char name[32];
+ int modify_counter;
+ int card_number;
+ int port_number;
+ char handle[32];
+ int magic; /* Reserved */
+ int enabled; /* Reserved */
+ int caps;
+#define MIXER_CAP_VIRTUAL 0x00000001
+#define MIXER_CAP_LAYOUT_B 0x00000002 /* For internal use only */
+#define MIXER_CAP_NARROW 0x00000004 /* Conserve horiz space */
+ int flags; /* Reserved */
+ int nrext;
+ /*
+ * The priority field can be used to select the default (motherboard)
+ * mixer device. The mixer with the highest priority is the
+ * most preferred one. -2 or less means that this device cannot be used
+ * as the default mixer.
+ */
+ int priority;
+ oss_devnode_t devnode; /* Device special file name (absolute path) */
+ int legacy_device;
+ int filler[245]; /* Reserved */
+} oss_mixerinfo;
+
+typedef struct oss_midi_info
+{
+ int dev; /* Midi device number */
+ char name[64];
+ int busy; /* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */
+ int pid;
+ char cmd[64]; /* Command using the device (if known) */
+ int caps;
+#define MIDI_CAP_MPU401 0x00000001 /**** OBSOLETE ****/
+#define MIDI_CAP_INPUT 0x00000002
+#define MIDI_CAP_OUTPUT 0x00000004
+#define MIDI_CAP_INOUT (MIDI_CAP_INPUT|MIDI_CAP_OUTPUT)
+#define MIDI_CAP_VIRTUAL 0x00000008 /* Pseudo device */
+#define MIDI_CAP_MTCINPUT 0x00000010 /* Supports SNDCTL_MIDI_MTCINPUT */
+#define MIDI_CAP_CLIENT 0x00000020 /* Virtual client side device */
+#define MIDI_CAP_SERVER 0x00000040 /* Virtual server side device */
+#define MIDI_CAP_INTERNAL 0x00000080 /* Internal (synth) device */
+#define MIDI_CAP_EXTERNAL 0x00000100 /* external (MIDI port) device */
+#define MIDI_CAP_PTOP 0x00000200 /* Point to point link to one device */
+#define MIDI_CAP_MTC 0x00000400 /* MTC/SMPTE (control) device */
+ int magic; /* Reserved for internal use */
+ int card_number;
+ int port_number;
+ int enabled; /* 1=enabled, 0=device not ready at this moment */
+ int flags; /* For internal use only - no practical meaning */
+ char handle[32];
+ oss_longname_t song_name; /* Song name (if known) */
+ oss_label_t label; /* Device label (if given) */
+ int latency; /* In usecs, -1=unknown */
+ oss_devnode_t devnode; /* Device special file name (absolute path) */
+ int legacy_device; /* Legacy device mapping */
+ int filler[235];
+} oss_midi_info;
+
+typedef struct oss_card_info
+{
+ int card;
+ char shortname[16];
+ char longname[128];
+ int flags;
+ char hw_info[400];
+ int intr_count, ack_count;
+ int filler[154];
+} oss_card_info;
+
+#define SNDCTL_SYSINFO __SIOR ('X', 1, oss_sysinfo)
+#define OSS_SYSINFO SNDCTL_SYSINFO /* Old name */
+
+#define SNDCTL_MIX_NRMIX __SIOR ('X', 2, int)
+#define SNDCTL_MIX_NREXT __SIOWR('X', 3, int)
+#define SNDCTL_MIX_EXTINFO __SIOWR('X', 4, oss_mixext)
+#define SNDCTL_MIX_READ __SIOWR('X', 5, oss_mixer_value)
+#define SNDCTL_MIX_WRITE __SIOWR('X', 6, oss_mixer_value)
+
+#define SNDCTL_AUDIOINFO __SIOWR('X', 7, oss_audioinfo)
+#define SNDCTL_MIX_ENUMINFO __SIOWR('X', 8, oss_mixer_enuminfo)
+#define SNDCTL_MIDIINFO __SIOWR('X', 9, oss_midi_info)
+#define SNDCTL_MIXERINFO __SIOWR('X',10, oss_mixerinfo)
+#define SNDCTL_CARDINFO __SIOWR('X',11, oss_card_info)
+#define SNDCTL_ENGINEINFO __SIOWR('X',12, oss_audioinfo)
+#define SNDCTL_AUDIOINFO_EX __SIOWR('X',13, oss_audioinfo)
+
+#define SNDCTL_MIX_DESCRIPTION __SIOWR('X',14, oss_mixer_enuminfo)
+
+/* ioctl codes 'X', 200-255 are reserved for internal use */
+
+/*
+ * Few more "globally" available ioctl calls.
+ */
+#define SNDCTL_SETSONG __SIOW ('Y', 2, oss_longname_t)
+#define SNDCTL_GETSONG __SIOR ('Y', 2, oss_longname_t)
+#define SNDCTL_SETNAME __SIOW ('Y', 3, oss_longname_t)
+#define SNDCTL_SETLABEL __SIOW ('Y', 4, oss_label_t)
+#define SNDCTL_GETLABEL __SIOR ('Y', 4, oss_label_t)
+/*
+ * The "new" mixer API definitions end here.
+ ***************************************
+ */
+
+/*
+ *********************************************************
+ * Few routines that are included in -lOSSlib
+ *
+ * At this moment this interface is not used. OSSlib contains just
+ * stubs that call the related system calls directly.
+ */
+#ifdef OSSLIB
+extern int osslib_open (const char *path, int flags, int dummy);
+extern void osslib_close (int fd);
+extern int osslib_write (int fd, const void *buf, int count);
+extern int osslib_read (int fd, void *buf, int count);
+extern int osslib_ioctl (int fd, unsigned int request, void *arg);
+#else
+# define osslib_open open
+# define osslib_close close
+# define osslib_write write
+# define osslib_read read
+# define osslib_ioctl ioctl
+#endif
+
+#if 1
+#define SNDCTL_DSP_NONBLOCK __SIO ('P',14) /* Obsolete. Not supported any more */
+#endif
+
+#if 1
+/*
+ * Some obsolete macros that are not part of Open Sound System API.
+ */
+#define SOUND_PCM_READ_RATE SOUND_PCM_READ_RATE_is_obsolete
+#define SOUND_PCM_READ_BITS SOUND_PCM_READ_BITS_is_obsolete
+#define SOUND_PCM_READ_CHANNELS SOUND_PCM_READ_CHANNELS_is_obsolete
+#define SOUND_PCM_WRITE_RATE SOUND_PCM_WRITE_RATE_is_obsolet_use_SNDCTL_DSP_SPEED_instead
+#define SOUND_PCM_WRITE_CHANNELS SOUND_PCM_WRITE_CHANNELS_is_obsolete_use_SNDCTL_DSP_CHANNELS_instead
+#define SOUND_PCM_WRITE_BITS SOUND_PCM_WRITE_BITS_is_obsolete_use_SNDCTL_DSP_SETFMT_instead
+#define SOUND_PCM_POST SOUND_PCM_POST_is_obsolete_use_SNDCTL_DSP_POST_instead
+#define SOUND_PCM_RESET SOUND_PCM_RESET_is_obsolete_use_SNDCTL_DSP_HALT_instead
+#define SOUND_PCM_SYNC SOUND_PCM_SYNC_is_obsolete_use_SNDCTL_DSP_SYNC_instead
+#define SOUND_PCM_SUBDIVIDE SOUND_PCM_SUBDIVIDE_is_obsolete_use_SNDCTL_DSP_SUBDIVIDE_instead
+#define SOUND_PCM_SETFRAGMENT SOUND_PCM_SETFRAGMENT_is_obsolete_use_SNDCTL_DSP_SETFRAGMENT_instead
+#define SOUND_PCM_GETFMTS SOUND_PCM_GETFMTS_is_obsolete_use_SNDCTL_DSP_GETFMTS_instead
+#define SOUND_PCM_SETFMT SOUND_PCM_SETFMT_is_obsolete_use_SNDCTL_DSP_SETFMT_instead
+#define SOUND_PCM_GETOSPACE SOUND_PCM_GETOSPACE_is_obsolete_use_SNDCTL_DSP_GETOSPACE_instead
+#define SOUND_PCM_GETISPACE SOUND_PCM_GETISPACE_is_obsolete_use_SNDCTL_DSP_GETISPACE_instead
+#define SOUND_PCM_NONBLOCK SOUND_PCM_NONBLOCK_is_obsolete_use_SNDCTL_DSP_NONBLOCK_instead
+#define SOUND_PCM_GETCAPS SOUND_PCM_GETCAPS_is_obsolete_use_SNDCTL_DSP_GETCAPS_instead
+#define SOUND_PCM_GETTRIGGER SOUND_PCM_GETTRIGGER_is_obsolete_use_SNDCTL_DSP_GETTRIGGER_instead
+#define SOUND_PCM_SETTRIGGER SOUND_PCM_SETTRIGGER_is_obsolete_use_SNDCTL_DSP_SETTRIGGER_instead
+#define SOUND_PCM_SETSYNCRO SOUND_PCM_SETSYNCRO_is_obsolete_use_SNDCTL_DSP_SETSYNCRO_instead
+#define SOUND_PCM_GETIPTR SOUND_PCM_GETIPTR_is_obsolete_use_SNDCTL_DSP_GETIPTR_instead
+#define SOUND_PCM_GETOPTR SOUND_PCM_GETOPTR_is_obsolete_use_SNDCTL_DSP_GETOPTR_instead
+#define SOUND_PCM_MAPINBUF SOUND_PCM_MAPINBUF_is_obsolete_use_SNDCTL_DSP_MAPINBUF_instead
+#define SOUND_PCM_MAPOUTBUF SOUND_PCM_MAPOUTBUF_is_obsolete_use_SNDCTL_DSP_MAPOUTBUF_instead
+#endif
+
+#endif
--- /dev/null
+/* GStreamer OSS4 audio source
+ * 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.
+ */
+
+/**
+ * SECTION:element-oss4src
+ * @short_description: record sound from your sound card using OSS4
+ *
+ * <refsect2>
+ * <para>
+ * This element lets you record sound using the Open Sound System (OSS)
+ * version 4.
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * <programlisting>
+ * gst-launch -v oss4src ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
+ * </programlisting>
+ * will record sound from your sound card using OSS4 and encode it to an
+ * Ogg/Vorbis file (this will only work if your mixer settings are right
+ * and the right inputs areenabled etc.)
+ * </para>
+ * </refsect2>
+ *
+ * Since: 0.10.7
+ */
+
+/* FIXME: make sure we're not doing ioctls from the app thread (e.g. via the
+ * mixer interface) while recording */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+
+#include <gst/interfaces/mixer.h>
+#include <gst/gst-i18n-plugin.h>
+
+#define NO_LEGACY_MIXER
+#include "oss4-audio.h"
+#include "oss4-source.h"
+#include "oss4-property-probe.h"
+#include "oss4-soundcard.h"
+
+#define GST_OSS4_SOURCE_IS_OPEN(src) (GST_OSS4_SOURCE(src)->fd != -1)
+
+GST_DEBUG_CATEGORY_EXTERN (oss4src_debug);
+#define GST_CAT_DEFAULT oss4src_debug
+
+#define DEFAULT_DEVICE NULL
+#define DEFAULT_DEVICE_NAME NULL
+
+enum
+{
+ PROP_0,
+ PROP_DEVICE,
+ PROP_DEVICE_NAME
+};
+
+static void gst_oss4_source_init_interfaces (GType type);
+
+GST_BOILERPLATE_FULL (GstOss4Source, gst_oss4_source, GstAudioSrc,
+ GST_TYPE_AUDIO_SRC, gst_oss4_source_init_interfaces);
+
+static void gst_oss4_source_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void gst_oss4_source_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+
+static void gst_oss4_source_dispose (GObject * object);
+static void gst_oss4_source_finalize (GstOss4Source * osssrc);
+
+static GstCaps *gst_oss4_source_getcaps (GstBaseSrc * bsrc);
+
+static gboolean gst_oss4_source_open (GstAudioSrc * asrc,
+ gboolean silent_errors);
+static gboolean gst_oss4_source_open_func (GstAudioSrc * asrc);
+static gboolean gst_oss4_source_close (GstAudioSrc * asrc);
+static gboolean gst_oss4_source_prepare (GstAudioSrc * asrc,
+ GstRingBufferSpec * spec);
+static gboolean gst_oss4_source_unprepare (GstAudioSrc * asrc);
+static guint gst_oss4_source_read (GstAudioSrc * asrc, gpointer data,
+ guint length);
+static guint gst_oss4_source_delay (GstAudioSrc * asrc);
+static void gst_oss4_source_reset (GstAudioSrc * asrc);
+
+static void
+gst_oss4_source_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+ GstPadTemplate *templ;
+
+ gst_element_class_set_details_simple (element_class,
+ "OSS v4 Audio Source", "Source/Audio",
+ "Capture from a sound card via OSS version 4",
+ "Tim-Philipp Müller <tim centricular net>");
+
+ templ = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+ gst_oss4_audio_get_template_caps ());
+ gst_element_class_add_pad_template (element_class, templ);
+}
+static void
+gst_oss4_source_class_init (GstOss4SourceClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseSrcClass *gstbasesrc_class;
+ GstBaseAudioSrcClass *gstbaseaudiosrc_class;
+ GstAudioSrcClass *gstaudiosrc_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasesrc_class = (GstBaseSrcClass *) klass;
+ gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
+ gstaudiosrc_class = (GstAudioSrcClass *) klass;
+
+ gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss4_source_dispose);
+ gobject_class->finalize =
+ (GObjectFinalizeFunc) GST_DEBUG_FUNCPTR (gst_oss4_source_finalize);
+ gobject_class->get_property =
+ GST_DEBUG_FUNCPTR (gst_oss4_source_get_property);
+ gobject_class->set_property =
+ GST_DEBUG_FUNCPTR (gst_oss4_source_set_property);
+
+ gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss4_source_getcaps);
+
+ gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss4_source_open_func);
+ gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss4_source_prepare);
+ gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss4_source_unprepare);
+ gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss4_source_close);
+ gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss4_source_read);
+ gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss4_source_delay);
+ gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss4_source_reset);
+
+ g_object_class_install_property (gobject_class, PROP_DEVICE,
+ g_param_spec_string ("device", "Device",
+ "OSS4 device (e.g. /dev/oss/hdaudio0/pcm0 or /dev/dspN) "
+ "(NULL = use first available device)",
+ DEFAULT_DEVICE, G_PARAM_READWRITE));
+
+ g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
+ g_param_spec_string ("device-name", "Device name",
+ "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
+ G_PARAM_READABLE));
+}
+
+static void
+gst_oss4_source_init (GstOss4Source * osssrc, GstOss4SourceClass * g_class)
+{
+ const gchar *device;
+
+ device = g_getenv ("AUDIODEV");
+ if (device == NULL)
+ device = DEFAULT_DEVICE;
+
+ osssrc->fd = -1;
+ osssrc->device = g_strdup (device);
+ osssrc->device_name = g_strdup (DEFAULT_DEVICE_NAME);
+ osssrc->device_name = NULL;
+}
+
+static void
+gst_oss4_source_finalize (GstOss4Source * oss)
+{
+ g_free (oss->device);
+ oss->device = NULL;
+
+ g_list_free (oss->property_probe_list);
+ oss->property_probe_list = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (oss));
+}
+
+static void
+gst_oss4_source_dispose (GObject * object)
+{
+ GstOss4Source *oss = GST_OSS4_SOURCE (object);
+
+ if (oss->probed_caps) {
+ gst_caps_unref (oss->probed_caps);
+ oss->probed_caps = NULL;
+ }
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_oss4_source_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstOss4Source *oss;
+
+ oss = GST_OSS4_SOURCE (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ GST_OBJECT_LOCK (oss);
+ if (oss->fd == -1) {
+ g_free (oss->device);
+ oss->device = g_value_dup_string (value);
+ g_free (oss->device_name);
+ oss->device_name = NULL;
+ } else {
+ g_warning ("%s: can't change \"device\" property while audio source "
+ "is open", GST_OBJECT_NAME (oss));
+ }
+ GST_OBJECT_UNLOCK (oss);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_oss4_source_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstOss4Source *oss;
+
+ oss = GST_OSS4_SOURCE (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ GST_OBJECT_LOCK (oss);
+ g_value_set_string (value, oss->device);
+ GST_OBJECT_UNLOCK (oss);
+ break;
+ case PROP_DEVICE_NAME:
+ GST_OBJECT_LOCK (oss);
+ /* If device is set, try to retrieve the name even if we're not open */
+ if (oss->fd == -1 && oss->device != NULL) {
+ if (gst_oss4_source_open (GST_AUDIO_SRC (oss), TRUE)) {
+ g_value_set_string (value, oss->device_name);
+ gst_oss4_source_close (GST_AUDIO_SRC (oss));
+ } else {
+ g_value_set_string (value, NULL);
+ }
+ } else {
+ g_value_set_string (value, oss->device_name);
+ }
+ GST_OBJECT_UNLOCK (oss);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstCaps *
+gst_oss4_source_getcaps (GstBaseSrc * bsrc)
+{
+ GstOss4Source *oss;
+ GstCaps *caps;
+
+ oss = GST_OSS4_SOURCE (bsrc);
+
+ if (oss->fd == -1) {
+ caps = gst_caps_copy (gst_oss4_audio_get_template_caps ());
+ } else if (oss->probed_caps) {
+ caps = gst_caps_copy (oss->probed_caps);
+ } else {
+ caps = gst_oss4_audio_probe_caps (GST_OBJECT (oss), oss->fd);
+ if (caps != NULL && !gst_caps_is_empty (caps)) {
+ oss->probed_caps = gst_caps_copy (caps);
+ }
+ }
+
+ return caps;
+}
+
+/* note: we must not take the object lock here unless we fix up get_property */
+static gboolean
+gst_oss4_source_open (GstAudioSrc * asrc, gboolean silent_errors)
+{
+ GstOss4Source *oss;
+ gchar *device;
+ int mode;
+
+ oss = GST_OSS4_SOURCE (asrc);
+
+ if (oss->device)
+ device = g_strdup (oss->device);
+ else
+ device = gst_oss4_audio_find_device (GST_OBJECT_CAST (oss));
+
+ /* desperate times, desperate measures */
+ if (device == NULL)
+ device = g_strdup ("/dev/dsp0");
+
+ GST_INFO_OBJECT (oss, "Trying to open OSS4 device '%s'", device);
+
+ /* we open in non-blocking mode even if we don't really want to do writes
+ * non-blocking because we can't be sure that this is really a genuine
+ * OSS4 device with well-behaved drivers etc. We really don't want to
+ * hang forever under any circumstances. */
+ oss->fd = open (device, O_RDONLY | O_NONBLOCK, 0);
+ if (oss->fd == -1) {
+ switch (errno) {
+ case EBUSY:
+ goto busy;
+ case EACCES:
+ goto no_permission;
+ default:
+ goto open_failed;
+ }
+ }
+
+ GST_INFO_OBJECT (oss, "Opened device");
+
+ /* Make sure it's OSS4. If it's old OSS, let osssink handle it */
+ if (!gst_oss4_audio_check_version (GST_OBJECT_CAST (oss), oss->fd))
+ goto legacy_oss;
+
+ /* now remove the non-blocking flag. */
+ mode = fcntl (oss->fd, F_GETFL);
+ mode &= ~O_NONBLOCK;
+ if (fcntl (oss->fd, F_SETFL, mode) < 0) {
+ /* some drivers do no support unsetting the non-blocking flag, try to
+ * close/open the device then. This is racy but we error out properly. */
+ GST_WARNING_OBJECT (oss, "failed to unset O_NONBLOCK (buggy driver?), "
+ "will try to re-open device now");
+ gst_oss4_source_close (asrc);
+ if ((oss->fd = open (device, O_RDONLY, 0)) == -1)
+ goto non_block;
+ }
+
+ oss->open_device = device;
+
+ /* not using ENGINEINFO here because it sometimes returns a different and
+ * less useful name than AUDIOINFO for the same device */
+ if (!gst_oss4_property_probe_find_device_name (GST_OBJECT (oss), oss->fd,
+ oss->open_device, &oss->device_name)) {
+ oss->device_name = NULL;
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+busy:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, BUSY,
+ (_("Could not open audio device for playback. "
+ "Device is being used by another application.")), (NULL));
+ }
+ g_free (device);
+ return FALSE;
+ }
+no_permission:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+ (_("Could not open audio device for playback."
+ "You don't have permission to open the device.")),
+ GST_ERROR_SYSTEM);
+ }
+ g_free (device);
+ return FALSE;
+ }
+open_failed:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+ (_("Could not open audio device for playback.")), GST_ERROR_SYSTEM);
+ }
+ g_free (device);
+ return FALSE;
+ }
+legacy_oss:
+ {
+ gst_oss4_source_close (asrc);
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+ (_("Could not open audio device for playback."
+ "This version of the Open Sound System is not supported by this "
+ "element.")), ("Try the 'osssink' element instead"));
+ }
+ g_free (device);
+ return FALSE;
+ }
+non_block:
+ {
+ if (!silent_errors) {
+ GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
+ ("Unable to set device %s into non-blocking mode: %s",
+ oss->device, g_strerror (errno)));
+ }
+ g_free (device);
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_oss4_source_open_func (GstAudioSrc * asrc)
+{
+ return gst_oss4_source_open (asrc, FALSE);
+}
+
+static void
+gst_oss4_source_free_mixer_tracks (GstOss4Source * oss)
+{
+ g_list_foreach (oss->tracks, (GFunc) g_object_unref, NULL);
+ g_list_free (oss->tracks);
+ oss->tracks = NULL;
+}
+
+static gboolean
+gst_oss4_source_close (GstAudioSrc * asrc)
+{
+ GstOss4Source *oss;
+
+ oss = GST_OSS4_SOURCE (asrc);
+
+ if (oss->fd != -1) {
+ GST_DEBUG_OBJECT (oss, "closing device");
+ close (oss->fd);
+ oss->fd = -1;
+ }
+
+ oss->bytes_per_sample = 0;
+
+ gst_caps_replace (&oss->probed_caps, NULL);
+
+ g_free (oss->open_device);
+ oss->open_device = NULL;
+
+ g_free (oss->device_name);
+ oss->device_name = NULL;
+
+ gst_oss4_source_free_mixer_tracks (oss);
+
+ return TRUE;
+}
+
+static gboolean
+gst_oss4_source_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
+{
+ GstOss4Source *oss;
+
+ oss = GST_OSS4_SOURCE (asrc);
+
+ if (!gst_oss4_audio_set_format (GST_OBJECT_CAST (oss), oss->fd, spec)) {
+ GST_WARNING_OBJECT (oss, "Couldn't set requested format %" GST_PTR_FORMAT,
+ spec->caps);
+ return FALSE;
+ }
+
+ oss->bytes_per_sample = spec->bytes_per_sample;
+ return TRUE;
+}
+
+static gboolean
+gst_oss4_source_unprepare (GstAudioSrc * asrc)
+{
+ /* could do a SNDCTL_DSP_HALT, but the OSS manual recommends a close/open,
+ * since HALT won't properly reset some devices, apparently */
+
+ if (!gst_oss4_source_close (asrc))
+ goto couldnt_close;
+
+ if (!gst_oss4_source_open_func (asrc))
+ goto couldnt_reopen;
+
+ return TRUE;
+
+ /* ERRORS */
+couldnt_close:
+ {
+ GST_DEBUG_OBJECT (asrc, "Couldn't close the audio device");
+ return FALSE;
+ }
+couldnt_reopen:
+ {
+ GST_DEBUG_OBJECT (asrc, "Couldn't reopen the audio device");
+ return FALSE;
+ }
+}
+
+static guint
+gst_oss4_source_read (GstAudioSrc * asrc, gpointer data, guint length)
+{
+ GstOss4Source *oss;
+ int n;
+
+ oss = GST_OSS4_SOURCE_CAST (asrc);
+
+ n = read (oss->fd, data, length);
+ GST_LOG_OBJECT (asrc, "%u bytes, %u samples", n, n / oss->bytes_per_sample);
+
+ if (G_UNLIKELY (n < 0)) {
+ switch (errno) {
+ case ENOTSUP:
+ case EACCES:{
+ /* This is the most likely cause, I think */
+ GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
+ (_("Recording is not supported by this audio device.")),
+ ("read: %s (device: %s) (maybe this is an output-only device?)",
+ g_strerror (errno), oss->open_device));
+ break;
+ }
+ default:{
+ GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
+ (_("Error recording from audio device.")),
+ ("read: %s (device: %s)", g_strerror (errno), oss->open_device));
+ break;
+ }
+ }
+ }
+
+ return (guint) n;
+}
+
+static guint
+gst_oss4_source_delay (GstAudioSrc * asrc)
+{
+ audio_buf_info info = { 0, };
+ GstOss4Source *oss;
+ guint delay;
+
+ oss = GST_OSS4_SOURCE_CAST (asrc);
+
+ if (ioctl (oss->fd, SNDCTL_DSP_GETISPACE, &info) == -1) {
+ GST_LOG_OBJECT (oss, "GETISPACE failed: %s", g_strerror (errno));
+ return 0;
+ }
+
+ delay = (info.fragstotal * info.fragsize) - info.bytes;
+ GST_LOG_OBJECT (oss, "fragstotal:%d, fragsize:%d, bytes:%d, delay:%d");
+ return delay;
+}
+
+static void
+gst_oss4_source_reset (GstAudioSrc * asrc)
+{
+ /* There's nothing we can do here really: OSS can't handle access to the
+ * same device/fd from multiple threads and might deadlock or blow up in
+ * other ways if we try an ioctl SNDCTL_DSP_HALT or similar */
+}
+
+/* GstMixer interface, which we abuse here for input selection, because we
+ * don't have a proper interface for that and because that's what
+ * gnome-sound-recorder does. */
+
+/* GstMixerTrack is a plain GObject, so let's just use the GLib macro here */
+G_DEFINE_TYPE (GstOss4SourceInput, gst_oss4_source_input, GST_TYPE_MIXER_TRACK);
+
+static void
+gst_oss4_source_input_class_init (GstOss4SourceInputClass * klass)
+{
+ /* nothing to do here */
+}
+
+static void
+gst_oss4_source_input_init (GstOss4SourceInput * i)
+{
+ /* nothing to do here */
+}
+
+#if 0
+
+static void
+gst_ossmixer_ensure_track_list (GstOssMixer * mixer)
+{
+ gint i, master = -1;
+
+ g_return_if_fail (mixer->fd != -1);
+
+ if (mixer->tracklist)
+ return;
+
+ /* find master volume */
+ if (mixer->devmask & SOUND_MASK_VOLUME)
+ master = SOUND_MIXER_VOLUME;
+ else if (mixer->devmask & SOUND_MASK_PCM)
+ master = SOUND_MIXER_PCM;
+ else if (mixer->devmask & SOUND_MASK_SPEAKER)
+ master = SOUND_MIXER_SPEAKER; /* doubtful... */
+ /* else: no master, so we won't set any */
+
+ /* build track list */
+ for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
+ if (mixer->devmask & (1 << i)) {
+ GstMixerTrack *track;
+ gboolean input = FALSE, stereo = FALSE, record = FALSE;
+
+ /* track exists, make up capabilities */
+ if (MASK_BIT_IS_SET (mixer->stereomask, i))
+ stereo = TRUE;
+ if (MASK_BIT_IS_SET (mixer->recmask, i))
+ input = TRUE;
+ if (MASK_BIT_IS_SET (mixer->recdevs, i))
+ record = TRUE;
+
+ /* do we want mixer in our list? */
+ if (!((mixer->dir & GST_OSS_MIXER_CAPTURE && input == TRUE) ||
+ (mixer->dir & GST_OSS_MIXER_PLAYBACK && i != SOUND_MIXER_PCM)))
+ /* the PLAYBACK case seems hacky, but that's how 0.8 had it */
+ continue;
+
+ /* add track to list */
+ track = gst_ossmixer_track_new (mixer->fd, i, stereo ? 2 : 1,
+ (record ? GST_MIXER_TRACK_RECORD : 0) |
+ (input ? GST_MIXER_TRACK_INPUT :
+ GST_MIXER_TRACK_OUTPUT) |
+ ((master != i) ? 0 : GST_MIXER_TRACK_MASTER));
+ mixer->tracklist = g_list_append (mixer->tracklist, track);
+ }
+ }
+}
+
+/* unused with G_DISABLE_* */
+static G_GNUC_UNUSED gboolean
+gst_ossmixer_contains_track (GstOssMixer * mixer, GstOssMixerTrack * osstrack)
+{
+ const GList *item;
+
+ for (item = mixer->tracklist; item != NULL; item = item->next)
+ if (item->data == osstrack)
+ return TRUE;
+
+ return FALSE;
+}
+
+const GList *
+gst_ossmixer_list_tracks (GstOssMixer * mixer)
+{
+ gst_ossmixer_ensure_track_list (mixer);
+
+ return (const GList *) mixer->tracklist;
+}
+
+void
+gst_ossmixer_get_volume (GstOssMixer * mixer,
+ GstMixerTrack * track, gint * volumes)
+{
+ gint volume;
+ GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
+
+ g_return_if_fail (mixer->fd != -1);
+ g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
+
+ if (track->flags & GST_MIXER_TRACK_MUTE) {
+ volumes[0] = osstrack->lvol;
+ if (track->num_channels == 2) {
+ volumes[1] = osstrack->rvol;
+ }
+ } else {
+ /* get */
+ if (ioctl (mixer->fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
+ g_warning ("Error getting recording device (%d) volume: %s",
+ osstrack->track_num, g_strerror (errno));
+ volume = 0;
+ }
+
+ osstrack->lvol = volumes[0] = (volume & 0xff);
+ if (track->num_channels == 2) {
+ osstrack->rvol = volumes[1] = ((volume >> 8) & 0xff);
+ }
+ }
+}
+
+void
+gst_ossmixer_set_mute (GstOssMixer * mixer, GstMixerTrack * track,
+ gboolean mute)
+{
+ int volume;
+ GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
+
+ g_return_if_fail (mixer->fd != -1);
+ g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
+
+ if (mute) {
+ volume = 0;
+ } else {
+ volume = (osstrack->lvol & 0xff);
+ if (MASK_BIT_IS_SET (mixer->stereomask, osstrack->track_num)) {
+ volume |= ((osstrack->rvol & 0xff) << 8);
+ }
+ }
+
+ if (ioctl (mixer->fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
+ g_warning ("Error setting mixer recording device volume (0x%x): %s",
+ volume, g_strerror (errno));
+ return;
+ }
+
+ if (mute) {
+ track->flags |= GST_MIXER_TRACK_MUTE;
+ } else {
+ track->flags &= ~GST_MIXER_TRACK_MUTE;
+ }
+}
+#endif
+
+static gint
+gst_oss4_source_mixer_get_current_input (GstOss4Source * oss)
+{
+ int cur = -1;
+
+ if (ioctl (oss->fd, SNDCTL_DSP_GET_RECSRC, &cur) == -1 || cur < 0)
+ return -1;
+
+ return cur;
+}
+
+static const gchar *
+gst_oss4_source_mixer_update_record_flags (GstOss4Source * oss, gint cur_route)
+{
+ const gchar *cur_name = "";
+ GList *t;
+
+ for (t = oss->tracks; t != NULL; t = t->next) {
+ GstMixerTrack *track = t->data;
+
+ if (GST_OSS4_SOURCE_INPUT (track)->route == cur_route) {
+ if (!GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) {
+ track->flags |= GST_MIXER_TRACK_RECORD;
+ /* no point in sending a mixer-record-changes message here */
+ }
+ cur_name = track->label;
+ } else {
+ if (GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) {
+ track->flags &= ~GST_MIXER_TRACK_RECORD;
+ /* no point in sending a mixer-record-changes message here */
+ }
+ }
+ }
+
+ return cur_name;
+}
+
+static const GList *
+gst_oss4_source_mixer_list_tracks (GstMixer * mixer)
+{
+ oss_mixer_enuminfo names = { 0, };
+ GstOss4Source *oss;
+ const gchar *cur_name;
+ GList *tracks = NULL;
+ gint i, cur;
+
+ g_return_val_if_fail (mixer != NULL, NULL);
+ g_return_val_if_fail (GST_IS_OSS4_SOURCE (mixer), NULL);
+ g_return_val_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer), NULL);
+
+ oss = GST_OSS4_SOURCE (mixer);
+
+ if (oss->tracks != NULL && oss->tracks_static)
+ goto done;
+
+ if (ioctl (oss->fd, SNDCTL_DSP_GET_RECSRC_NAMES, &names) == -1)
+ goto get_recsrc_names_error;
+
+ oss->tracks_static = (names.version == 0);
+
+ GST_INFO_OBJECT (oss, "%d inputs (list is static: %s):", names.nvalues,
+ (oss->tracks_static) ? "yes" : "no");
+
+ for (i = 0; i < MIN (names.nvalues, OSS_ENUM_MAXVALUE + 1); ++i) {
+ GstMixerTrack *track;
+
+ track = g_object_new (GST_TYPE_OSS4_SOURCE_INPUT, NULL);
+ track->label = g_strdup (&names.strings[names.strindex[i]]);
+ track->flags = GST_MIXER_TRACK_INPUT;
+ track->num_channels = 2;
+ track->min_volume = 0;
+ track->max_volume = 100;
+ GST_OSS4_SOURCE_INPUT (track)->route = i;
+
+ GST_INFO_OBJECT (oss, " [%d] %s", i, track->label);
+ tracks = g_list_append (tracks, track);
+ }
+
+ gst_oss4_source_free_mixer_tracks (oss);
+ oss->tracks = tracks;
+
+done:
+
+ /* update RECORD flags */
+ cur = gst_oss4_source_mixer_get_current_input (oss);
+ cur_name = gst_oss4_source_mixer_update_record_flags (oss, cur);
+ GST_DEBUG_OBJECT (oss, "current input route: %d (%s)", cur, cur_name);
+
+ return (const GList *) oss->tracks;
+
+/* ERRORS */
+get_recsrc_names_error:
+ {
+ GST_WARNING_OBJECT (oss, "GET_RECSRC_NAMES failed: %s", g_strerror (errno));
+ return NULL;
+ }
+}
+
+static void
+gst_oss4_source_mixer_set_volume (GstMixer * mixer, GstMixerTrack * track,
+ gint * volumes)
+{
+ GstOss4Source *oss;
+ int new_vol, cur;
+
+ g_return_if_fail (mixer != NULL);
+ g_return_if_fail (track != NULL);
+ g_return_if_fail (GST_IS_MIXER_TRACK (track));
+ g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
+ g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
+
+ oss = GST_OSS4_SOURCE (mixer);
+
+ cur = gst_oss4_source_mixer_get_current_input (oss);
+ if (cur != GST_OSS4_SOURCE_INPUT (track)->route) {
+ GST_DEBUG_OBJECT (oss, "track not selected input route, ignoring request");
+ return;
+ }
+
+ new_vol = (volumes[1] << 8) | volumes[0];
+ if (ioctl (oss->fd, SNDCTL_DSP_SETRECVOL, &new_vol) == -1) {
+ GST_WARNING_OBJECT (oss, "SETRECVOL failed: %s", g_strerror (errno));
+ }
+}
+
+static void
+gst_oss4_source_mixer_get_volume (GstMixer * mixer, GstMixerTrack * track,
+ gint * volumes)
+{
+ GstOss4Source *oss;
+ int cur;
+
+ g_return_if_fail (mixer != NULL);
+ g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
+ g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
+
+ oss = GST_OSS4_SOURCE (mixer);
+
+ cur = gst_oss4_source_mixer_get_current_input (oss);
+ if (cur != GST_OSS4_SOURCE_INPUT (track)->route) {
+ volumes[0] = 0;
+ volumes[1] = 0;
+ } else {
+ int vol = -1;
+
+ if (ioctl (oss->fd, SNDCTL_DSP_GETRECVOL, &vol) == -1 || vol < 0) {
+ GST_WARNING_OBJECT (oss, "GETRECVOL failed: %s", g_strerror (errno));
+ volumes[0] = 100;
+ volumes[1] = 100;
+ } else {
+ volumes[0] = MIN (100, vol & 0xff);
+ volumes[1] = MIN (100, (vol >> 8) & 0xff);
+ }
+ }
+}
+
+static void
+gst_oss4_source_mixer_set_record (GstMixer * mixer, GstMixerTrack * track,
+ gboolean record)
+{
+ GstOss4Source *oss;
+ const gchar *cur_name;
+ gint cur;
+
+ g_return_if_fail (mixer != NULL);
+ g_return_if_fail (track != NULL);
+ g_return_if_fail (GST_IS_MIXER_TRACK (track));
+ g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
+ g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
+
+ oss = GST_OSS4_SOURCE (mixer);
+
+ cur = gst_oss4_source_mixer_get_current_input (oss);
+
+ /* stop recording for an input that's not selected anyway => nothing to do */
+ if (!record && cur != GST_OSS4_SOURCE_INPUT (track)->route)
+ goto done;
+
+ /* select recording for an input that's already selected => nothing to do
+ * (or should we mess with the recording volume in this case maybe?) */
+ if (record && cur == GST_OSS4_SOURCE_INPUT (track)->route)
+ goto done;
+
+ /* make current input stop recording: we can't really make an input stop
+ * recording, we can only select an input FOR recording, so we'll just ignore
+ * all requests to stop for now */
+ if (!record) {
+ GST_WARNING_OBJECT (oss, "Can't un-select an input as such, only switch "
+ "to a different input source");
+ /* FIXME: set recording volume to 0 maybe? */
+ } else {
+ int new_route = GST_OSS4_SOURCE_INPUT (track)->route;
+
+ /* select this input for recording */
+
+ if (ioctl (oss->fd, SNDCTL_DSP_SET_RECSRC, &new_route) == -1) {
+ GST_WARNING_OBJECT (oss, "Could not select input %d for recording: %s",
+ new_route, g_strerror (errno));
+ } else {
+ cur = new_route;
+ }
+ }
+
+done:
+
+ cur_name = gst_oss4_source_mixer_update_record_flags (oss, cur);
+ GST_DEBUG_OBJECT (oss, "active input route: %d (%s)", cur, cur_name);
+}
+
+static void
+gst_oss4_source_mixer_set_mute (GstMixer * mixer, GstMixerTrack * track,
+ gboolean mute)
+{
+ GstOss4Source *oss;
+
+ g_return_if_fail (mixer != NULL);
+ g_return_if_fail (track != NULL);
+ g_return_if_fail (GST_IS_MIXER_TRACK (track));
+ g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
+ g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
+
+ oss = GST_OSS4_SOURCE (mixer);
+
+ /* FIXME: implement gst_oss4_source_mixer_set_mute() - what to do here? */
+ /* oss4_mixer_set_mute (mixer->mixer, track, mute); */
+}
+
+static void
+gst_oss4_source_mixer_interface_init (GstMixerClass * klass)
+{
+ GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE;
+
+ klass->list_tracks = gst_oss4_source_mixer_list_tracks;
+ klass->set_volume = gst_oss4_source_mixer_set_volume;
+ klass->get_volume = gst_oss4_source_mixer_get_volume;
+ klass->set_mute = gst_oss4_source_mixer_set_mute;
+ klass->set_record = gst_oss4_source_mixer_set_record;
+}
+
+/* Implement the horror that is GstImplementsInterface */
+
+static gboolean
+gst_oss4_source_mixer_supported (GstImplementsInterface * iface,
+ GType iface_type)
+{
+ GstOss4Source *oss;
+ gboolean is_open;
+
+ g_return_val_if_fail (GST_IS_OSS4_SOURCE (iface), FALSE);
+ g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
+
+ oss = GST_OSS4_SOURCE (iface);
+
+ GST_OBJECT_LOCK (oss);
+ is_open = GST_OSS4_SOURCE_IS_OPEN (iface);
+ GST_OBJECT_UNLOCK (oss);
+
+ return is_open;
+}
+
+static void
+gst_oss4_source_mixer_implements_interface_init (GstImplementsInterfaceClass *
+ klass)
+{
+ klass->supported = gst_oss4_source_mixer_supported;
+}
+
+static void
+gst_oss4_source_init_interfaces (GType type)
+{
+ static const GInterfaceInfo implements_iface_info = {
+ (GInterfaceInitFunc) gst_oss4_source_mixer_implements_interface_init,
+ NULL,
+ NULL,
+ };
+ static const GInterfaceInfo mixer_iface_info = {
+ (GInterfaceInitFunc) gst_oss4_source_mixer_interface_init,
+ NULL,
+ NULL,
+ };
+
+ g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
+ &implements_iface_info);
+ g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
+
+ gst_oss4_add_property_probe_interface (type);
+}
--- /dev/null
+/* GStreamer OSS4 audio source
+ * 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.
+ */
+
+#ifndef GST_OSS4_SOURCE_H
+#define GST_OSS4_SOURCE_H
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiosrc.h>
+#include <gst/interfaces/mixertrack.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_OSS4_SOURCE (gst_oss4_source_get_type())
+#define GST_OSS4_SOURCE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_SOURCE,GstOss4Source))
+#define GST_OSS4_SOURCE_CAST(obj) ((GstOss4Source *)(obj))
+#define GST_OSS4_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_SOURCE,GstOss4SourceClass))
+#define GST_IS_OSS4_SOURCE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_SOURCE))
+#define GST_IS_OSS4_SOURCE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_SOURCE))
+
+typedef struct _GstOss4Source GstOss4Source;
+typedef struct _GstOss4SourceClass GstOss4SourceClass;
+
+struct _GstOss4Source {
+ GstAudioSrc audiosrc;
+
+ gchar * device; /* NULL if none was set */
+ gchar * open_device; /* the device we opened */
+ gchar * device_name; /* set if the device is open */
+ gint fd; /* -1 if not open */
+ gint bytes_per_sample;
+
+ GstCaps * probed_caps;
+
+ /* property probe interface */
+ GList * property_probe_list;
+
+ /* mixer interface */
+ GList * tracks;
+ gboolean tracks_static; /* FALSE if the list of inputs may change */
+};
+
+struct _GstOss4SourceClass {
+ GstAudioSrcClass audiosrc_class;
+};
+
+GType gst_oss4_source_get_type (void);
+
+/* our mixer track for input selection */
+#define GST_TYPE_OSS4_SOURCE_INPUT (gst_oss4_source_input_get_type())
+#define GST_OSS4_SOURCE_INPUT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS4_SOURCE_INPUT,GstOss4SourceInput))
+#define GST_OSS4_SOURCE_INPUT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS4_SOURCE_INPUT,GstOss4SourceInputClass))
+#define GST_IS_OSS4_SOURCE_INPUT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS4_SOURCE_INPUT))
+#define GST_IS_OSS4_SOURCE_INPUT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS4_SOURCE_INPUT))
+
+typedef struct _GstOss4SourceInput GstOss4SourceInput;
+typedef struct _GstOss4SourceInputClass GstOss4SourceInputClass;
+
+struct _GstOss4SourceInput {
+ GstMixerTrack mixer_track;
+
+ int route; /* number for SNDCTL_DSP_SET_RECSRC etc. */
+};
+
+struct _GstOss4SourceInputClass {
+ GstMixerTrackClass mixer_track_class;
+};
+
+GType gst_oss4_source_input_get_type (void);
+
+G_END_DECLS
+
+#endif /* GST_OSS4_SOURCE_H */
+
projects / platform / upstream / gst-plugins-good.git / commitdiff