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., 59 Temple Place - Suite 330,
20 * Boston, MA 02111-1307, USA.
24 * SECTION:element-osssrc
26 * This element lets you record sound using the Open Sound System (OSS).
29 * <title>Example pipelines</title>
31 * gst-launch -v osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
32 * ]| will record sound from your sound card using OSS and encode it to an
33 * Ogg/Vorbis file (this will only work if your mixer settings are right
34 * and the right inputs enabled etc.)
42 #include <sys/ioctl.h>
48 #ifdef HAVE_OSS_INCLUDE_IN_SYS
49 # include <sys/soundcard.h>
51 # ifdef HAVE_OSS_INCLUDE_IN_ROOT
52 # include <soundcard.h>
54 # ifdef HAVE_OSS_INCLUDE_IN_MACHINE
55 # include <machine/soundcard.h>
57 # error "What to include?"
58 # endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
59 # endif /* HAVE_OSS_INCLUDE_IN_ROOT */
60 #endif /* HAVE_OSS_INCLUDE_IN_SYS */
62 #include "gstosssrc.h"
65 #include <gst/gst-i18n-plugin.h>
67 GST_DEBUG_CATEGORY_EXTERN (oss_debug);
68 #define GST_CAT_DEFAULT oss_debug
70 static const GstElementDetails gst_oss_src_details =
71 GST_ELEMENT_DETAILS ("Audio Source (OSS)",
73 "Capture from a sound card via OSS",
74 "Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>");
76 #define DEFAULT_DEVICE "/dev/dsp"
77 #define DEFAULT_DEVICE_NAME ""
86 GST_BOILERPLATE_WITH_INTERFACE (GstOssSrc, gst_oss_src, GstAudioSrc,
87 GST_TYPE_AUDIO_SRC, GstMixer, GST_TYPE_MIXER, gst_oss_src_mixer);
89 GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssSrc, gst_oss_src_mixer);
91 static void gst_oss_src_get_property (GObject * object, guint prop_id,
92 GValue * value, GParamSpec * pspec);
93 static void gst_oss_src_set_property (GObject * object, guint prop_id,
94 const GValue * value, GParamSpec * pspec);
96 static void gst_oss_src_dispose (GObject * object);
97 static void gst_oss_src_finalize (GstOssSrc * osssrc);
99 static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc);
101 static gboolean gst_oss_src_open (GstAudioSrc * asrc);
102 static gboolean gst_oss_src_close (GstAudioSrc * asrc);
103 static gboolean gst_oss_src_prepare (GstAudioSrc * asrc,
104 GstRingBufferSpec * spec);
105 static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc);
106 static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length);
107 static guint gst_oss_src_delay (GstAudioSrc * asrc);
108 static void gst_oss_src_reset (GstAudioSrc * asrc);
112 static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
115 GST_STATIC_CAPS ("audio/x-raw-int, "
116 "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
117 "signed = (boolean) { TRUE, FALSE }, "
120 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
122 "signed = (boolean) { TRUE, FALSE }, "
125 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
130 gst_oss_src_dispose (GObject * object)
132 G_OBJECT_CLASS (parent_class)->dispose (object);
136 gst_oss_src_base_init (gpointer g_class)
138 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
140 gst_element_class_set_details (element_class, &gst_oss_src_details);
142 gst_element_class_add_pad_template (element_class,
143 gst_static_pad_template_get (&osssrc_src_factory));
147 gst_oss_src_class_init (GstOssSrcClass * klass)
149 GObjectClass *gobject_class;
150 GstElementClass *gstelement_class;
151 GstBaseSrcClass *gstbasesrc_class;
152 GstBaseAudioSrcClass *gstbaseaudiosrc_class;
153 GstAudioSrcClass *gstaudiosrc_class;
155 gobject_class = (GObjectClass *) klass;
156 gstelement_class = (GstElementClass *) klass;
157 gstbasesrc_class = (GstBaseSrcClass *) klass;
158 gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
159 gstaudiosrc_class = (GstAudioSrcClass *) klass;
161 gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss_src_dispose);
162 gobject_class->finalize =
163 (GObjectFinalizeFunc) GST_DEBUG_FUNCPTR (gst_oss_src_finalize);
164 gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_oss_src_get_property);
165 gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_oss_src_set_property);
167 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_src_getcaps);
169 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss_src_open);
170 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_src_prepare);
171 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_src_unprepare);
172 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss_src_close);
173 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss_src_read);
174 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss_src_delay);
175 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss_src_reset);
177 g_object_class_install_property (gobject_class, PROP_DEVICE,
178 g_param_spec_string ("device", "Device",
179 "OSS device (usually /dev/dspN)", DEFAULT_DEVICE, G_PARAM_READWRITE));
181 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
182 g_param_spec_string ("device-name", "Device name",
183 "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
188 gst_oss_src_set_property (GObject * object, guint prop_id,
189 const GValue * value, GParamSpec * pspec)
193 src = GST_OSS_SRC (object);
198 g_free (src->device);
199 src->device = g_value_dup_string (value);
202 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
208 gst_oss_src_get_property (GObject * object, guint prop_id,
209 GValue * value, GParamSpec * pspec)
213 src = GST_OSS_SRC (object);
217 g_value_set_string (value, src->device);
219 case PROP_DEVICE_NAME:
220 g_value_set_string (value, src->device_name);
223 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
229 gst_oss_src_init (GstOssSrc * osssrc, GstOssSrcClass * g_class)
233 GST_DEBUG ("initializing osssrc");
235 device = g_getenv ("AUDIODEV");
237 device = DEFAULT_DEVICE;
240 osssrc->device = g_strdup (device);
241 osssrc->device_name = g_strdup (DEFAULT_DEVICE_NAME);
242 osssrc->probed_caps = NULL;
246 gst_oss_src_finalize (GstOssSrc * osssrc)
248 g_free (osssrc->device);
249 g_free (osssrc->device_name);
251 G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (osssrc));
255 gst_oss_src_getcaps (GstBaseSrc * bsrc)
260 osssrc = GST_OSS_SRC (bsrc);
262 if (osssrc->fd == -1) {
263 GST_DEBUG_OBJECT (osssrc, "device not open, using template caps");
264 return NULL; /* base class will get template caps for us */
267 if (osssrc->probed_caps) {
268 GST_LOG_OBJECT (osssrc, "Returning cached caps");
269 return gst_caps_ref (osssrc->probed_caps);
272 caps = gst_oss_helper_probe_caps (osssrc->fd);
275 osssrc->probed_caps = gst_caps_ref (caps);
278 GST_INFO_OBJECT (osssrc, "returning caps %" GST_PTR_FORMAT, caps);
286 /* well... hacker's delight explains... */
292 x = x - ((x >> 1) & 0x55555555);
293 x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
294 x = (x + (x >> 4)) & 0x0f0f0f0f;
297 return (x & 0x0000003f) - 1;
301 gst_oss_src_get_format (GstBufferFormat fmt)
307 result = AFMT_MU_LAW;
313 result = AFMT_IMA_ADPCM;
319 result = AFMT_S16_LE;
322 result = AFMT_S16_BE;
328 result = AFMT_U16_LE;
331 result = AFMT_U16_BE;
344 gst_oss_src_open (GstAudioSrc * asrc)
349 oss = GST_OSS_SRC (asrc);
354 oss->fd = open (oss->device, mode, 0);
365 oss->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_CAPTURE);
368 g_free (oss->device_name);
369 oss->device_name = g_strdup (oss->mixer->cardname);
376 GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
377 (_("Could not open audio device for recording. "
378 "You don't have permission to open the device.")),
384 GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
385 (_("Could not open audio device for recording.")),
386 ("Unable to open device %s for recording: %s",
387 oss->device, g_strerror (errno)));
393 gst_oss_src_close (GstAudioSrc * asrc)
397 oss = GST_OSS_SRC (asrc);
402 gst_ossmixer_free (oss->mixer);
406 gst_caps_replace (&oss->probed_caps, NULL);
412 gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
415 struct audio_buf_info info;
419 oss = GST_OSS_SRC (asrc);
421 mode = fcntl (oss->fd, F_GETFL);
423 if (fcntl (oss->fd, F_SETFL, mode) == -1)
426 fmt = gst_oss_src_get_format (spec->format);
430 tmp = ilog2 (spec->segsize);
431 tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
432 GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",
433 spec->segsize, spec->segtotal, tmp);
435 SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT");
437 SET_PARAM (oss, SNDCTL_DSP_RESET, 0, "RESET");
439 SET_PARAM (oss, SNDCTL_DSP_SETFMT, fmt, "SETFMT");
440 if (spec->channels == 2)
441 SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
442 SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS");
443 SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED");
445 GET_PARAM (oss, SNDCTL_DSP_GETISPACE, &info, "GETISPACE");
447 spec->segsize = info.fragsize;
448 spec->segtotal = info.fragstotal;
450 if (spec->width != 16 && spec->width != 8)
453 spec->bytes_per_sample = (spec->width / 8) * spec->channels;
454 oss->bytes_per_sample = (spec->width / 8) * spec->channels;
455 memset (spec->silence_sample, 0, spec->bytes_per_sample);
457 GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
458 spec->segsize, spec->segtotal, tmp);
464 GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
465 ("Unable to set device %s in non blocking mode: %s",
466 oss->device, g_strerror (errno)), (NULL));
471 GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
472 ("Unable to get format %d", spec->format), (NULL));
477 GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
478 ("Unexpected width %d", spec->width), (NULL));
484 gst_oss_src_unprepare (GstAudioSrc * asrc)
486 /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
488 if (!gst_oss_src_close (asrc))
491 if (!gst_oss_src_open (asrc))
498 GST_DEBUG_OBJECT (asrc, "Could not close the audio device");
503 GST_DEBUG_OBJECT (asrc, "Could not reopen the audio device");
509 gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length)
511 return read (GST_OSS_SRC (asrc)->fd, data, length);
515 gst_oss_src_delay (GstAudioSrc * asrc)
521 oss = GST_OSS_SRC (asrc);
523 #ifdef SNDCTL_DSP_GETODELAY
524 ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
531 ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
533 delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
535 return delay / oss->bytes_per_sample;
539 gst_oss_src_reset (GstAudioSrc * asrc)
541 /* There's nothing we can do here really: OSS can't handle access to the
542 * same device/fd from multiple threads and might deadlock or blow up in
543 * other ways if we try an ioctl SNDCTL_DSP_RESET or similar */