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-rw-r--r--sys/oss4/oss4-source.c1004
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diff --git a/sys/oss4/oss4-source.c b/sys/oss4/oss4-source.c
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+/* 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);
+}