2 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
3 * 2000,2005 Wim Taymans <wim@fluendo.com>
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
20 * Boston, MA 02110-1301, USA.
24 * SECTION:element-osssink
26 * This element lets you output sound using the Open Sound System (OSS).
28 * Note that you should almost always use generic audio conversion elements
29 * like audioconvert and audioresample in front of an audiosink to make sure
30 * your pipeline works under all circumstances (those conversion elements will
31 * act in passthrough-mode if no conversion is necessary).
34 * <title>Example pipelines</title>
36 * gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.1 ! osssink
37 * ]| will output a sine wave (continuous beep sound) to your sound card (with
38 * a very low volume as precaution).
40 * gst-launch-1.0 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! osssink
41 * ]| will play an Ogg/Vorbis audio file and output it using the Open Sound System.
48 #include <sys/ioctl.h>
54 #ifdef HAVE_OSS_INCLUDE_IN_SYS
55 # include <sys/soundcard.h>
57 # ifdef HAVE_OSS_INCLUDE_IN_ROOT
58 # include <soundcard.h>
60 # ifdef HAVE_OSS_INCLUDE_IN_MACHINE
61 # include <machine/soundcard.h>
63 # error "What to include?"
64 # endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
65 # endif /* HAVE_OSS_INCLUDE_IN_ROOT */
66 #endif /* HAVE_OSS_INCLUDE_IN_SYS */
69 #include "gstosssink.h"
71 #include <gst/gst-i18n-plugin.h>
73 GST_DEBUG_CATEGORY_EXTERN (oss_debug);
74 #define GST_CAT_DEFAULT oss_debug
76 static void gst_oss_sink_dispose (GObject * object);
77 static void gst_oss_sink_finalise (GObject * object);
79 static void gst_oss_sink_get_property (GObject * object, guint prop_id,
80 GValue * value, GParamSpec * pspec);
81 static void gst_oss_sink_set_property (GObject * object, guint prop_id,
82 const GValue * value, GParamSpec * pspec);
84 static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink, GstCaps * filter);
86 static gboolean gst_oss_sink_open (GstAudioSink * asink);
87 static gboolean gst_oss_sink_close (GstAudioSink * asink);
88 static gboolean gst_oss_sink_prepare (GstAudioSink * asink,
89 GstAudioRingBufferSpec * spec);
90 static gboolean gst_oss_sink_unprepare (GstAudioSink * asink);
91 static gint gst_oss_sink_write (GstAudioSink * asink, gpointer data,
93 static guint gst_oss_sink_delay (GstAudioSink * asink);
94 static void gst_oss_sink_reset (GstAudioSink * asink);
96 /* OssSink signals and args */
102 #define DEFAULT_DEVICE "/dev/dsp"
109 #define FORMATS "{" GST_AUDIO_NE(S16)","GST_AUDIO_NE(U16)", S8, U8 }"
111 static GstStaticPadTemplate osssink_sink_factory =
112 GST_STATIC_PAD_TEMPLATE ("sink",
115 GST_STATIC_CAPS ("audio/x-raw, "
116 "format = (string) " FORMATS ", "
117 "layout = (string) interleaved, "
118 "rate = (int) [ 1, MAX ], "
119 "channels = (int) 1; "
121 "format = (string) " FORMATS ", "
122 "layout = (string) interleaved, "
123 "rate = (int) [ 1, MAX ], "
124 "channels = (int) 2, " "channel-mask = (bitmask) 0x3")
127 /* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */
129 #define gst_oss_sink_parent_class parent_class
130 G_DEFINE_TYPE (GstOssSink, gst_oss_sink, GST_TYPE_AUDIO_SINK);
133 gst_oss_sink_dispose (GObject * object)
135 GstOssSink *osssink = GST_OSSSINK (object);
137 if (osssink->probed_caps) {
138 gst_caps_unref (osssink->probed_caps);
139 osssink->probed_caps = NULL;
142 G_OBJECT_CLASS (parent_class)->dispose (object);
146 gst_oss_sink_class_init (GstOssSinkClass * klass)
148 GObjectClass *gobject_class;
149 GstElementClass *gstelement_class;
150 GstBaseSinkClass *gstbasesink_class;
151 GstAudioSinkClass *gstaudiosink_class;
153 gobject_class = (GObjectClass *) klass;
154 gstelement_class = (GstElementClass *) klass;
155 gstbasesink_class = (GstBaseSinkClass *) klass;
156 gstaudiosink_class = (GstAudioSinkClass *) klass;
158 parent_class = g_type_class_peek_parent (klass);
160 gobject_class->dispose = gst_oss_sink_dispose;
161 gobject_class->finalize = gst_oss_sink_finalise;
162 gobject_class->get_property = gst_oss_sink_get_property;
163 gobject_class->set_property = gst_oss_sink_set_property;
165 g_object_class_install_property (gobject_class, PROP_DEVICE,
166 g_param_spec_string ("device", "Device",
167 "OSS device (usually /dev/dspN)", DEFAULT_DEVICE,
168 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
170 gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps);
172 gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open);
173 gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close);
174 gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare);
175 gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare);
176 gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write);
177 gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay);
178 gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset);
180 gst_element_class_set_static_metadata (gstelement_class, "Audio Sink (OSS)",
182 "Output to a sound card via OSS",
183 "Erik Walthinsen <omega@cse.ogi.edu>, "
184 "Wim Taymans <wim.taymans@chello.be>");
186 gst_element_class_add_static_pad_template (gstelement_class,
187 &osssink_sink_factory);
191 gst_oss_sink_init (GstOssSink * osssink)
195 GST_DEBUG_OBJECT (osssink, "initializing osssink");
197 device = g_getenv ("AUDIODEV");
199 device = DEFAULT_DEVICE;
200 osssink->device = g_strdup (device);
205 gst_oss_sink_finalise (GObject * object)
207 GstOssSink *osssink = GST_OSSSINK (object);
209 g_free (osssink->device);
211 G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (object));
215 gst_oss_sink_set_property (GObject * object, guint prop_id,
216 const GValue * value, GParamSpec * pspec)
220 sink = GST_OSSSINK (object);
224 g_free (sink->device);
225 sink->device = g_value_dup_string (value);
226 if (sink->probed_caps) {
227 gst_caps_unref (sink->probed_caps);
228 sink->probed_caps = NULL;
232 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
238 gst_oss_sink_get_property (GObject * object, guint prop_id,
239 GValue * value, GParamSpec * pspec)
243 sink = GST_OSSSINK (object);
247 g_value_set_string (value, sink->device);
250 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
256 gst_oss_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
261 osssink = GST_OSSSINK (bsink);
263 if (osssink->fd == -1) {
264 caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink));
265 } else if (osssink->probed_caps) {
266 caps = gst_caps_ref (osssink->probed_caps);
268 caps = gst_oss_helper_probe_caps (osssink->fd);
269 if (caps && !gst_caps_is_empty (caps)) {
270 osssink->probed_caps = gst_caps_ref (caps);
274 if (filter && caps) {
275 GstCaps *intersection;
278 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
279 gst_caps_unref (caps);
289 /* well... hacker's delight explains... */
295 x = x - ((x >> 1) & 0x55555555);
296 x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
297 x = (x + (x >> 4)) & 0x0f0f0f0f;
300 return (x & 0x0000003f) - 1;
304 gst_oss_sink_get_format (GstAudioRingBufferFormatType fmt, GstAudioFormat rfmt)
309 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
310 result = AFMT_MU_LAW;
312 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
315 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM:
316 result = AFMT_IMA_ADPCM;
318 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
321 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
324 case GST_AUDIO_FORMAT_U8:
327 case GST_AUDIO_FORMAT_S16LE:
328 result = AFMT_S16_LE;
330 case GST_AUDIO_FORMAT_S16BE:
331 result = AFMT_S16_BE;
333 case GST_AUDIO_FORMAT_S8:
336 case GST_AUDIO_FORMAT_U16LE:
337 result = AFMT_U16_LE;
339 case GST_AUDIO_FORMAT_U16BE:
340 result = AFMT_U16_BE;
356 gst_oss_sink_open (GstAudioSink * asink)
361 oss = GST_OSSSINK (asink);
366 oss->fd = open (oss->device, mode, 0);
383 GST_ELEMENT_ERROR (oss, RESOURCE, BUSY,
384 (_("Could not open audio device for playback. "
385 "Device is being used by another application.")), (NULL));
390 GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
391 (_("Could not open audio device for playback. "
392 "You don't have permission to open the device.")),
398 GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
399 (_("Could not open audio device for playback.")), GST_ERROR_SYSTEM);
405 gst_oss_sink_close (GstAudioSink * asink)
407 close (GST_OSSSINK (asink)->fd);
408 GST_OSSSINK (asink)->fd = -1;
413 gst_oss_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
416 struct audio_buf_info info;
419 guint width, rate, channels;
421 oss = GST_OSSSINK (asink);
423 /* we opened non-blocking so that we can detect if the device is available
424 * without hanging forever. We now want to remove the non-blocking flag. */
425 mode = fcntl (oss->fd, F_GETFL);
427 if (fcntl (oss->fd, F_SETFL, mode) == -1) {
428 /* some drivers do no support unsetting the non-blocking flag, try to
429 * close/open the device then. This is racy but we error out properly. */
430 gst_oss_sink_close (asink);
431 if ((oss->fd = open (oss->device, O_WRONLY, 0)) == -1)
435 tmp = gst_oss_sink_get_format (spec->type,
436 GST_AUDIO_INFO_FORMAT (&spec->info));
440 width = GST_AUDIO_INFO_WIDTH (&spec->info);
441 rate = GST_AUDIO_INFO_RATE (&spec->info);
442 channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
444 if (width != 16 && width != 8)
447 SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp, "SETFMT");
449 SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
450 SET_PARAM (oss, SNDCTL_DSP_CHANNELS, channels, "CHANNELS");
451 SET_PARAM (oss, SNDCTL_DSP_SPEED, rate, "SPEED");
453 tmp = ilog2 (spec->segsize);
454 tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
455 GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",
456 spec->segsize, spec->segtotal, tmp);
458 SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT");
459 GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info, "GETOSPACE");
461 spec->segsize = info.fragsize;
462 spec->segtotal = info.fragstotal;
464 oss->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
466 GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
467 spec->segsize, spec->segtotal, tmp);
474 GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
475 ("Unable to set device %s in non blocking mode: %s",
476 oss->device, g_strerror (errno)));
481 GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
482 ("Unable to get format (%d, %d)", spec->type,
483 GST_AUDIO_INFO_FORMAT (&spec->info)));
488 GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
489 ("unexpected width %d", width));
495 gst_oss_sink_unprepare (GstAudioSink * asink)
497 /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
499 if (!gst_oss_sink_close (asink))
502 if (!gst_oss_sink_open (asink))
510 GST_DEBUG_OBJECT (asink, "Could not close the audio device");
515 GST_DEBUG_OBJECT (asink, "Could not reopen the audio device");
521 gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length)
523 return write (GST_OSSSINK (asink)->fd, data, length);
527 gst_oss_sink_delay (GstAudioSink * asink)
533 oss = GST_OSSSINK (asink);
535 #ifdef SNDCTL_DSP_GETODELAY
536 ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
543 ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
545 delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
547 return delay / oss->bytes_per_sample;
551 gst_oss_sink_reset (GstAudioSink * asink)
553 /* There's nothing we can do here really: OSS can't handle access to the
554 * same device/fd from multiple threads and might deadlock or blow up in
555 * other ways if we try an ioctl SNDCTL_DSP_RESET or similar */