/* GStreamer OSS4 audio source * Copyright (C) 2007-2008 Tim-Philipp Müller * * 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 * * This element lets you record sound using the Open Sound System (OSS) * version 4. * * * Example pipelines * |[ * gst-launch -v oss4src ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg * ]| 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.) * * * 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 #include #include #include #include #include #include #include #include #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_static_metadata (element_class, "OSS v4 Audio Source", "Source/Audio", "Capture from a sound card via OSS version 4", "Tim-Philipp Müller "); 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; GstBaseSrcClass *gstbasesrc_class; GstAudioSrcClass *gstaudiosrc_class; gobject_class = (GObjectClass *) klass; gstbasesrc_class = (GstBaseSrcClass *) klass; gstaudiosrc_class = (GstAudioSrcClass *) klass; gobject_class->dispose = gst_oss4_source_dispose; gobject_class->finalize = (GObjectFinalizeFunc) gst_oss4_source_finalize; gobject_class->get_property = gst_oss4_source_get_property; gobject_class->set_property = 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_PARAM_STATIC_STRINGS)); 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 | G_PARAM_STATIC_STRINGS)); } 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 { gchar *name = NULL; gst_oss4_property_probe_find_device_name_nofd (GST_OBJECT (oss), oss->device, &name); g_value_set_string (value, name); g_free (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_source_getcaps (GstBaseSrc * bsrc) { GstOss4Source *oss; GstCaps *caps; oss = GST_OSS4_SOURCE (bsrc); if (oss->fd == -1) { caps = 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", info.fragstotal, info.fragsize, info.bytes, delay); 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) { 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)); /* 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 (GstMixerInterface * iface) { GST_MIXER_TYPE (iface) = GST_MIXER_HARDWARE; iface->list_tracks = gst_oss4_source_mixer_list_tracks; iface->set_volume = gst_oss4_source_mixer_set_volume; iface->get_volume = gst_oss4_source_mixer_get_volume; iface->set_mute = gst_oss4_source_mixer_set_mute; iface->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); }