/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2000,2005 Wim Taymans * * gstosssink.c: * * 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-osssink * @short_description: output sound using OSS * * * * This element lets you output sound using the Open Sound System (OSS). * * * 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). * * Example pipelines * * * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! osssink * * will output a sine wave (continuous beep sound) to your sound card (with * a very low volume as precaution). * * * * gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! osssink * * will play an Ogg/Vorbis audio file and output it using the Open Sound System. * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #ifdef HAVE_OSS_INCLUDE_IN_SYS # include #else # ifdef HAVE_OSS_INCLUDE_IN_ROOT # include # else # ifdef HAVE_OSS_INCLUDE_IN_MACHINE # include # else # error "What to include?" # endif /* HAVE_OSS_INCLUDE_IN_MACHINE */ # endif /* HAVE_OSS_INCLUDE_IN_ROOT */ #endif /* HAVE_OSS_INCLUDE_IN_SYS */ #include "common.h" #include "gstosssink.h" #include GST_DEBUG_CATEGORY_EXTERN (oss_debug); #define GST_CAT_DEFAULT oss_debug /* elementfactory information */ static const GstElementDetails gst_oss_sink_details = GST_ELEMENT_DETAILS ("Audio Sink (OSS)", "Sink/Audio", "Output to a sound card via OSS", "Erik Walthinsen , " "Wim Taymans "); static void gst_oss_sink_base_init (gpointer g_class); static void gst_oss_sink_class_init (GstOssSinkClass * klass); static void gst_oss_sink_init (GstOssSink * osssink); static void gst_oss_sink_dispose (GObject * object); static void gst_oss_sink_finalise (GObject * object); static void gst_oss_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_oss_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink); static gboolean gst_oss_sink_open (GstAudioSink * asink); static gboolean gst_oss_sink_close (GstAudioSink * asink); static gboolean gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec); static gboolean gst_oss_sink_unprepare (GstAudioSink * asink); static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length); static guint gst_oss_sink_delay (GstAudioSink * asink); static void gst_oss_sink_reset (GstAudioSink * asink); /* OssSink signals and args */ enum { LAST_SIGNAL }; #define DEFAULT_DEVICE "/dev/dsp" enum { PROP_0, PROP_DEVICE, }; static GstStaticPadTemplate osssink_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, " "signed = (boolean) { TRUE, FALSE }, " "width = (int) 16, " "depth = (int) 16, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; " "audio/x-raw-int, " "signed = (boolean) { TRUE, FALSE }, " "width = (int) 8, " "depth = (int) 8, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]") ); static GstElementClass *parent_class = NULL; /* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */ GType gst_oss_sink_get_type (void) { static GType osssink_type = 0; if (!osssink_type) { static const GTypeInfo osssink_info = { sizeof (GstOssSinkClass), gst_oss_sink_base_init, NULL, (GClassInitFunc) gst_oss_sink_class_init, NULL, NULL, sizeof (GstOssSink), 0, (GInstanceInitFunc) gst_oss_sink_init, }; osssink_type = g_type_register_static (GST_TYPE_AUDIO_SINK, "GstOssSink", &osssink_info, 0); } return osssink_type; } static void gst_oss_sink_dispose (GObject * object) { GstOssSink *osssink = GST_OSSSINK (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_oss_sink_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_set_details (element_class, &gst_oss_sink_details); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&osssink_sink_factory)); } static void gst_oss_sink_class_init (GstOssSinkClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseSinkClass *gstbasesink_class; GstBaseAudioSinkClass *gstbaseaudiosink_class; GstAudioSinkClass *gstaudiosink_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasesink_class = (GstBaseSinkClass *) klass; gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; gstaudiosink_class = (GstAudioSinkClass *) klass; parent_class = g_type_class_peek_parent (klass); gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss_sink_dispose); gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_oss_sink_finalise); gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_oss_sink_get_property); gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_oss_sink_set_property); g_object_class_install_property (gobject_class, PROP_DEVICE, g_param_spec_string ("device", "Device", "OSS device (usually /dev/dspN)", DEFAULT_DEVICE, G_PARAM_READWRITE)); gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps); gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open); gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close); gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare); gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare); gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write); gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay); gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset); } static void gst_oss_sink_init (GstOssSink * osssink) { const gchar *device; GST_DEBUG_OBJECT (osssink, "initializing osssink"); device = g_getenv ("AUDIODEV"); if (device == NULL) device = DEFAULT_DEVICE; osssink->device = g_strdup (device); osssink->fd = -1; } static void gst_oss_sink_finalise (GObject * object) { GstOssSink *osssink = GST_OSSSINK (object); g_free (osssink->device); G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (object)); } static void gst_oss_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstOssSink *sink; sink = GST_OSSSINK (object); switch (prop_id) { case PROP_DEVICE: g_free (sink->device); sink->device = g_value_dup_string (value); if (sink->probed_caps) { gst_caps_unref (sink->probed_caps); sink->probed_caps = NULL; } break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_oss_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstOssSink *sink; sink = GST_OSSSINK (object); switch (prop_id) { case PROP_DEVICE: g_value_set_string (value, sink->device); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstCaps * gst_oss_sink_getcaps (GstBaseSink * bsink) { GstOssSink *osssink; GstCaps *caps; osssink = GST_OSSSINK (bsink); if (osssink->fd == -1) { caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink))); } else if (osssink->probed_caps) { caps = gst_caps_copy (osssink->probed_caps); } else { caps = gst_oss_helper_probe_caps (osssink->fd); if (caps && !gst_caps_is_empty (caps)) { osssink->probed_caps = gst_caps_copy (caps); } } return caps; } static gint 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; } static gint gst_oss_sink_get_format (GstBufferFormat fmt) { gint result; switch (fmt) { case GST_MU_LAW: result = AFMT_MU_LAW; break; case GST_A_LAW: result = AFMT_A_LAW; break; case GST_IMA_ADPCM: result = AFMT_IMA_ADPCM; break; case GST_U8: result = AFMT_U8; break; case GST_S16_LE: result = AFMT_S16_LE; break; case GST_S16_BE: result = AFMT_S16_BE; break; case GST_S8: result = AFMT_S8; break; case GST_U16_LE: result = AFMT_U16_LE; break; case GST_U16_BE: result = AFMT_U16_BE; break; case GST_MPEG: result = AFMT_MPEG; break; default: result = 0; break; } return result; } static gboolean gst_oss_sink_open (GstAudioSink * asink) { GstOssSink *oss; int mode; oss = GST_OSSSINK (asink); mode = O_WRONLY; mode |= O_NONBLOCK; oss->fd = open (oss->device, mode, 0); if (oss->fd == -1) { switch (errno) { case EBUSY: goto busy; case EACCES: goto no_permission; default: goto open_failed; } } return TRUE; /* ERRORS */ busy: { GST_ELEMENT_ERROR (oss, RESOURCE, BUSY, (_("Could not open audio device for playback. " "Device is being used by another application.")), (NULL)); return FALSE; } no_permission: { 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); return FALSE; } open_failed: { GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE, (_("Could not open audio device for playback.")), GST_ERROR_SYSTEM); return FALSE; } } static gboolean gst_oss_sink_close (GstAudioSink * asink) { close (GST_OSSSINK (asink)->fd); GST_OSSSINK (asink)->fd = -1; return TRUE; } static gboolean gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec) { GstOssSink *oss; struct audio_buf_info info; int mode; int tmp; oss = GST_OSSSINK (asink); /* we opened non-blocking so that we can detect if the device is available * without hanging forever. We now want to remove the non-blocking flag. */ mode = fcntl (oss->fd, F_GETFL); mode &= ~O_NONBLOCK; if (fcntl (oss->fd, F_SETFL, mode) == -1) { /* 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_oss_sink_close (asink); if ((oss->fd = open (oss->device, O_WRONLY, 0)) == -1) goto non_block; } tmp = gst_oss_sink_get_format (spec->format); if (tmp == 0) goto wrong_format; if (spec->width != 16 && spec->width != 8) goto dodgy_width; SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp, "SETFMT"); if (spec->channels == 2) SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO"); SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS"); SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED"); tmp = ilog2 (spec->segsize); tmp = ((spec->segtotal & 0x7fff) << 16) | tmp; GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x", spec->segsize, spec->segtotal, tmp); SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT"); GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info, "GETOSPACE"); spec->segsize = info.fragsize; spec->segtotal = info.fragstotal; spec->bytes_per_sample = (spec->width / 8) * spec->channels; oss->bytes_per_sample = (spec->width / 8) * spec->channels; GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x", spec->segsize, spec->segtotal, tmp); return TRUE; /* ERRORS */ non_block: { GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL), ("Unable to set device %s in non blocking mode: %s", oss->device, g_strerror (errno))); return FALSE; } wrong_format: { GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL), ("Unable to get format %d", spec->format)); return FALSE; } dodgy_width: { GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL), ("unexpected width %d", spec->width)); return FALSE; } } static gboolean gst_oss_sink_unprepare (GstAudioSink * asink) { /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */ if (!gst_oss_sink_close (asink)) goto couldnt_close; if (!gst_oss_sink_open (asink)) goto couldnt_reopen; return TRUE; /* ERRORS */ couldnt_close: { GST_DEBUG_OBJECT (asink, "Could not close the audio device"); return FALSE; } couldnt_reopen: { GST_DEBUG_OBJECT (asink, "Could not reopen the audio device"); return FALSE; } } static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length) { return write (GST_OSSSINK (asink)->fd, data, length); } static guint gst_oss_sink_delay (GstAudioSink * asink) { GstOssSink *oss; gint delay = 0; gint ret; oss = GST_OSSSINK (asink); #ifdef SNDCTL_DSP_GETODELAY ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay); #else ret = -1; #endif if (ret < 0) { audio_buf_info info; ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info); delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes); } return delay / oss->bytes_per_sample; } static void gst_oss_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_RESET or similar */ }