2 * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
19 * Boston, MA 02111-1307, USA.
23 * SECTION:element-alsasrc
24 * @see_also: alsasink, alsamixer
26 * This element reads data from an audio card using the ALSA API.
29 * <title>Example pipelines</title>
31 * gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
32 * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis.
35 * Last reviewed on 2006-03-01 (0.10.4)
41 #include <sys/ioctl.h>
47 #include <alsa/asoundlib.h>
49 #include "gstalsasrc.h"
50 #include "gstalsadeviceprobe.h"
52 #include <gst/gst-i18n-plugin.h>
54 #define DEFAULT_PROP_DEVICE "default"
55 #define DEFAULT_PROP_DEVICE_NAME ""
56 #define DEFAULT_PROP_CARD_NAME ""
67 static void gst_alsasrc_init_interfaces (GType type);
68 #define gst_alsasrc_parent_class parent_class
69 G_DEFINE_TYPE_WITH_CODE (GstAlsaSrc, gst_alsasrc,
70 GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces (g_define_type_id));
72 GST_IMPLEMENT_ALSA_MIXER_METHODS (GstAlsaSrc, gst_alsasrc_mixer);
74 static void gst_alsasrc_finalize (GObject * object);
75 static void gst_alsasrc_set_property (GObject * object,
76 guint prop_id, const GValue * value, GParamSpec * pspec);
77 static void gst_alsasrc_get_property (GObject * object,
78 guint prop_id, GValue * value, GParamSpec * pspec);
80 static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter);
82 static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
83 static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
84 GstAudioRingBufferSpec * spec);
85 static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);
86 static gboolean gst_alsasrc_close (GstAudioSrc * asrc);
87 static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length);
88 static guint gst_alsasrc_delay (GstAudioSrc * asrc);
89 static void gst_alsasrc_reset (GstAudioSrc * asrc);
90 static GstStateChangeReturn gst_alsasrc_change_state (GstElement * element,
91 GstStateChange transition);
92 static GstFlowReturn gst_alsasrc_create (GstBaseSrc * bsrc, guint64 offset,
93 guint length, GstBuffer ** outbuf);
94 static GstClockTime gst_alsasrc_get_timestamp (GstAlsaSrc * src);
97 /* AlsaSrc signals and args */
103 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
104 # define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
106 # define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
109 static GstStaticPadTemplate alsasrc_src_factory =
110 GST_STATIC_PAD_TEMPLATE ("src",
113 GST_STATIC_CAPS ("audio/x-raw, "
114 "format = (string) " GST_AUDIO_FORMATS_ALL ", "
115 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
119 gst_alsasrc_finalize (GObject * object)
121 GstAlsaSrc *src = GST_ALSA_SRC (object);
123 g_free (src->device);
124 g_mutex_free (src->alsa_lock);
126 G_OBJECT_CLASS (parent_class)->finalize (object);
130 gst_alsasrc_init_interfaces (GType type)
132 static const GInterfaceInfo mixer_iface_info = {
133 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
138 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
140 gst_alsa_type_add_device_property_probe_interface (type);
144 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
146 GObjectClass *gobject_class;
147 GstElementClass *gstelement_class;
148 GstBaseSrcClass *gstbasesrc_class;
149 GstAudioSrcClass *gstaudiosrc_class;
151 gobject_class = (GObjectClass *) klass;
152 gstelement_class = (GstElementClass *) klass;
153 gstbasesrc_class = (GstBaseSrcClass *) klass;
154 gstaudiosrc_class = (GstAudioSrcClass *) klass;
156 gobject_class->finalize = gst_alsasrc_finalize;
157 gobject_class->get_property = gst_alsasrc_get_property;
158 gobject_class->set_property = gst_alsasrc_set_property;
160 gst_element_class_set_details_simple (gstelement_class,
161 "Audio source (ALSA)", "Source/Audio",
162 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
164 gst_element_class_add_pad_template (gstelement_class,
165 gst_static_pad_template_get (&alsasrc_src_factory));
167 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
169 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
170 gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_alsasrc_create);
172 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
173 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
174 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
175 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
176 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
177 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
178 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
180 g_object_class_install_property (gobject_class, PROP_DEVICE,
181 g_param_spec_string ("device", "Device",
182 "ALSA device, as defined in an asound configuration file",
183 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
185 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
186 g_param_spec_string ("device-name", "Device name",
187 "Human-readable name of the sound device",
188 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
190 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
191 g_param_spec_string ("card-name", "Card name",
192 "Human-readable name of the sound card",
193 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
197 gst_alsasrc_get_timestamp (GstAlsaSrc * src)
199 snd_pcm_status_t *status;
200 snd_htimestamp_t htstamp;
201 snd_timestamp_t tstamp;
202 GstClockTime timestamp;
203 snd_pcm_uframes_t availmax;
206 GST_DEBUG_OBJECT (src, "Getting alsa timestamp!");
209 GST_ERROR_OBJECT (src, "No alsa handle created yet !");
210 return GST_CLOCK_TIME_NONE;
213 if (snd_pcm_status_malloc (&status) != 0) {
214 GST_ERROR_OBJECT (src, "snd_pcm_status_malloc failed");
215 return GST_CLOCK_TIME_NONE;
218 if (snd_pcm_status (src->handle, status) != 0) {
219 GST_ERROR_OBJECT (src, "snd_pcm_status failed");
220 snd_pcm_status_free (status);
221 return GST_CLOCK_TIME_NONE;
224 /* get high resolution time stamp from driver */
225 snd_pcm_status_get_htstamp (status, &htstamp);
226 timestamp = GST_TIMESPEC_TO_TIME (htstamp);
227 if (timestamp == 0) {
228 GST_INFO_OBJECT (src,
229 "This alsa source does support high resolution timestamps");
230 snd_pcm_status_get_tstamp (status, &tstamp);
231 timestamp = GST_TIMEVAL_TO_TIME (tstamp);
232 if (timestamp == 0) {
233 GST_INFO_OBJECT (src,
234 "This alsa source does support low resolution timestamps");
235 timestamp = gst_util_get_timestamp ();
238 GST_DEBUG_OBJECT (src, "Base ts: %" GST_TIME_FORMAT,
239 GST_TIME_ARGS (timestamp));
240 if (timestamp == 0) {
241 /* This timestamp is supposed to represent the last sample, so 0 (which
242 can be returned on some ALSA setups (such as mine)) must mean that it
243 is invalid, unless there's just one sample, but we'll ignore that. */
244 GST_WARNING_OBJECT (src,
245 "No timestamp returned from snd_pcm_status_get_htstamp");
246 return GST_CLOCK_TIME_NONE;
249 /* Max available frames sets the depth of the buffer */
250 availmax = snd_pcm_status_get_avail_max (status);
252 /* Compensate the fact that the timestamp references the last sample */
253 offset = -gst_util_uint64_scale_int (availmax * 2, GST_SECOND, src->rate);
254 /* Compensate for the delay until the package is available */
255 offset += gst_util_uint64_scale_int (snd_pcm_status_get_delay (status),
256 GST_SECOND, src->rate);
258 snd_pcm_status_free (status);
260 /* just in case, should not happen */
261 if (-offset > timestamp)
266 /* Take first ts into account */
267 if (src->first_alsa_ts == GST_CLOCK_TIME_NONE) {
268 src->first_alsa_ts = timestamp;
270 timestamp -= src->first_alsa_ts;
272 GST_DEBUG_OBJECT (src, "ALSA timestamp : %" GST_TIME_FORMAT,
273 GST_TIME_ARGS (timestamp));
278 gst_alsasrc_set_property (GObject * object, guint prop_id,
279 const GValue * value, GParamSpec * pspec)
283 src = GST_ALSA_SRC (object);
287 g_free (src->device);
288 src->device = g_value_dup_string (value);
289 if (src->device == NULL) {
290 src->device = g_strdup (DEFAULT_PROP_DEVICE);
294 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
300 gst_alsasrc_get_property (GObject * object, guint prop_id,
301 GValue * value, GParamSpec * pspec)
305 src = GST_ALSA_SRC (object);
309 g_value_set_string (value, src->device);
311 case PROP_DEVICE_NAME:
312 g_value_take_string (value,
313 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
314 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
317 g_value_take_string (value,
318 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
319 src->device, SND_PCM_STREAM_CAPTURE));
322 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
327 static GstStateChangeReturn
328 gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
330 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
331 GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (element);
332 GstAlsaSrc *asrc = GST_ALSA_SRC (element);
335 switch (transition) {
336 /* Show the compiler that we care */
337 case GST_STATE_CHANGE_NULL_TO_READY:
338 case GST_STATE_CHANGE_READY_TO_PAUSED:
339 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
340 case GST_STATE_CHANGE_PAUSED_TO_READY:
341 case GST_STATE_CHANGE_READY_TO_NULL:
344 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
346 asrc->driver_timestamps = FALSE;
347 if (GST_IS_SYSTEM_CLOCK (clk)) {
349 g_object_get (clk, "clock-type", &clocktype, NULL);
350 if (clocktype == GST_CLOCK_TYPE_MONOTONIC) {
351 asrc->driver_timestamps = TRUE;
356 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
362 gst_alsasrc_create (GstBaseSrc * bsrc, guint64 offset, guint length,
365 GstFlowReturn ret = GST_FLOW_OK;
366 GstAlsaSrc *asrc = GST_ALSA_SRC (bsrc);
369 GST_BASE_SRC_CLASS (parent_class)->create (bsrc, offset, length, outbuf);
370 if (asrc->driver_timestamps == TRUE && *outbuf) {
371 GstClockTime ts = gst_alsasrc_get_timestamp (asrc);
372 if (GST_CLOCK_TIME_IS_VALID (ts)) {
373 GST_BUFFER_TIMESTAMP (*outbuf) = ts;
381 gst_alsasrc_init (GstAlsaSrc * alsasrc)
383 GST_DEBUG_OBJECT (alsasrc, "initializing");
385 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
386 alsasrc->cached_caps = NULL;
387 alsasrc->driver_timestamps = FALSE;
388 alsasrc->first_alsa_ts = GST_CLOCK_TIME_NONE;
390 alsasrc->alsa_lock = g_mutex_new ();
393 #define CHECK(call, error) \
395 if ((err = call) < 0) \
401 gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
403 GstElementClass *element_class;
404 GstPadTemplate *pad_template;
406 GstCaps *caps, *templ_caps;
408 src = GST_ALSA_SRC (bsrc);
410 if (src->handle == NULL) {
411 GST_DEBUG_OBJECT (src, "device not open, using template caps");
412 return GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
415 if (src->cached_caps) {
416 GST_LOG_OBJECT (src, "Returning cached caps");
418 return gst_caps_intersect_full (filter, src->cached_caps,
419 GST_CAPS_INTERSECT_FIRST);
421 return gst_caps_ref (src->cached_caps);
424 element_class = GST_ELEMENT_GET_CLASS (src);
425 pad_template = gst_element_class_get_pad_template (element_class, "src");
426 g_return_val_if_fail (pad_template != NULL, NULL);
428 templ_caps = gst_pad_template_get_caps (pad_template);
429 GST_INFO_OBJECT (src, "template caps %" GST_PTR_FORMAT, templ_caps);
431 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
433 gst_caps_unref (templ_caps);
436 src->cached_caps = gst_caps_ref (caps);
439 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
442 GstCaps *intersection;
445 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
446 gst_caps_unref (caps);
454 set_hwparams (GstAlsaSrc * alsa)
458 snd_pcm_hw_params_t *params;
460 snd_pcm_hw_params_malloc (¶ms);
462 /* choose all parameters */
463 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
464 /* set the interleaved read/write format */
465 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
467 /* set the sample format */
468 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
470 /* set the count of channels */
471 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
473 /* set the stream rate */
475 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
477 if (rrate != alsa->rate)
480 if (alsa->buffer_time != -1) {
481 /* set the buffer time */
482 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
483 &alsa->buffer_time, NULL), buffer_time);
485 if (alsa->period_time != -1) {
486 /* set the period time */
487 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
488 &alsa->period_time, NULL), period_time);
491 /* write the parameters to device */
492 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
494 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
497 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
500 snd_pcm_hw_params_free (params);
506 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
507 ("Broken configuration for recording: no configurations available: %s",
508 snd_strerror (err)));
509 snd_pcm_hw_params_free (params);
514 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
515 ("Access type not available for recording: %s", snd_strerror (err)));
516 snd_pcm_hw_params_free (params);
521 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
522 ("Sample format not available for recording: %s", snd_strerror (err)));
523 snd_pcm_hw_params_free (params);
530 if ((alsa->channels) == 1)
531 msg = g_strdup (_("Could not open device for recording in mono mode."));
532 if ((alsa->channels) == 2)
533 msg = g_strdup (_("Could not open device for recording in stereo mode."));
534 if ((alsa->channels) > 2)
537 ("Could not open device for recording in %d-channel mode"),
539 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
540 ("%s", snd_strerror (err)));
542 snd_pcm_hw_params_free (params);
547 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
548 ("Rate %iHz not available for recording: %s",
549 alsa->rate, snd_strerror (err)));
550 snd_pcm_hw_params_free (params);
555 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
556 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
557 snd_pcm_hw_params_free (params);
562 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
563 ("Unable to set buffer time %i for recording: %s",
564 alsa->buffer_time, snd_strerror (err)));
565 snd_pcm_hw_params_free (params);
570 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
571 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
572 snd_pcm_hw_params_free (params);
577 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
578 ("Unable to set period time %i for recording: %s", alsa->period_time,
579 snd_strerror (err)));
580 snd_pcm_hw_params_free (params);
585 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
586 ("Unable to get period size for recording: %s", snd_strerror (err)));
587 snd_pcm_hw_params_free (params);
592 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
593 ("Unable to set hw params for recording: %s", snd_strerror (err)));
594 snd_pcm_hw_params_free (params);
600 set_swparams (GstAlsaSrc * alsa)
603 snd_pcm_sw_params_t *params;
605 snd_pcm_sw_params_malloc (¶ms);
607 /* get the current swparams */
608 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
609 /* allow the transfer when at least period_size samples can be processed */
610 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
611 alsa->period_size), set_avail);
612 /* start the transfer on first read */
613 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
614 0), start_threshold);
616 #if GST_CHECK_ALSA_VERSION(1,0,16)
617 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
619 /* align all transfers to 1 sample */
620 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
623 /* write the parameters to the recording device */
624 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
626 snd_pcm_sw_params_free (params);
632 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
633 ("Unable to determine current swparams for playback: %s",
634 snd_strerror (err)));
635 snd_pcm_sw_params_free (params);
640 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
641 ("Unable to set start threshold mode for playback: %s",
642 snd_strerror (err)));
643 snd_pcm_sw_params_free (params);
648 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
649 ("Unable to set avail min for playback: %s", snd_strerror (err)));
650 snd_pcm_sw_params_free (params);
653 #if !GST_CHECK_ALSA_VERSION(1,0,16)
656 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
657 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
658 snd_pcm_sw_params_free (params);
664 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
665 ("Unable to set sw params for playback: %s", snd_strerror (err)));
666 snd_pcm_sw_params_free (params);
672 alsasrc_parse_spec (GstAlsaSrc * alsa, GstAudioRingBufferSpec * spec)
674 switch (spec->type) {
675 case GST_BUFTYPE_RAW:
676 switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
677 case GST_AUDIO_FORMAT_U8:
678 alsa->format = SND_PCM_FORMAT_U8;
680 case GST_AUDIO_FORMAT_S8:
681 alsa->format = SND_PCM_FORMAT_S8;
683 case GST_AUDIO_FORMAT_S16LE:
684 alsa->format = SND_PCM_FORMAT_S16_LE;
686 case GST_AUDIO_FORMAT_S16BE:
687 alsa->format = SND_PCM_FORMAT_S16_BE;
689 case GST_AUDIO_FORMAT_U16LE:
690 alsa->format = SND_PCM_FORMAT_U16_LE;
692 case GST_AUDIO_FORMAT_U16BE:
693 alsa->format = SND_PCM_FORMAT_U16_BE;
695 case GST_AUDIO_FORMAT_S24_32LE:
696 alsa->format = SND_PCM_FORMAT_S24_LE;
698 case GST_AUDIO_FORMAT_S24_32BE:
699 alsa->format = SND_PCM_FORMAT_S24_BE;
701 case GST_AUDIO_FORMAT_U24_32LE:
702 alsa->format = SND_PCM_FORMAT_U24_LE;
704 case GST_AUDIO_FORMAT_U24_32BE:
705 alsa->format = SND_PCM_FORMAT_U24_BE;
707 case GST_AUDIO_FORMAT_S32LE:
708 alsa->format = SND_PCM_FORMAT_S32_LE;
710 case GST_AUDIO_FORMAT_S32BE:
711 alsa->format = SND_PCM_FORMAT_S32_BE;
713 case GST_AUDIO_FORMAT_U32LE:
714 alsa->format = SND_PCM_FORMAT_U32_LE;
716 case GST_AUDIO_FORMAT_U32BE:
717 alsa->format = SND_PCM_FORMAT_U32_BE;
719 case GST_AUDIO_FORMAT_S24LE:
720 alsa->format = SND_PCM_FORMAT_S24_3LE;
722 case GST_AUDIO_FORMAT_S24BE:
723 alsa->format = SND_PCM_FORMAT_S24_3BE;
725 case GST_AUDIO_FORMAT_U24LE:
726 alsa->format = SND_PCM_FORMAT_U24_3LE;
728 case GST_AUDIO_FORMAT_U24BE:
729 alsa->format = SND_PCM_FORMAT_U24_3BE;
731 case GST_AUDIO_FORMAT_S20LE:
732 alsa->format = SND_PCM_FORMAT_S20_3LE;
734 case GST_AUDIO_FORMAT_S20BE:
735 alsa->format = SND_PCM_FORMAT_S20_3BE;
737 case GST_AUDIO_FORMAT_U20LE:
738 alsa->format = SND_PCM_FORMAT_U20_3LE;
740 case GST_AUDIO_FORMAT_U20BE:
741 alsa->format = SND_PCM_FORMAT_U20_3BE;
743 case GST_AUDIO_FORMAT_S18LE:
744 alsa->format = SND_PCM_FORMAT_S18_3LE;
746 case GST_AUDIO_FORMAT_S18BE:
747 alsa->format = SND_PCM_FORMAT_S18_3BE;
749 case GST_AUDIO_FORMAT_U18LE:
750 alsa->format = SND_PCM_FORMAT_U18_3LE;
752 case GST_AUDIO_FORMAT_U18BE:
753 alsa->format = SND_PCM_FORMAT_U18_3BE;
755 case GST_AUDIO_FORMAT_F32LE:
756 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
758 case GST_AUDIO_FORMAT_F32BE:
759 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
761 case GST_AUDIO_FORMAT_F64LE:
762 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
764 case GST_AUDIO_FORMAT_F64BE:
765 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
771 case GST_BUFTYPE_A_LAW:
772 alsa->format = SND_PCM_FORMAT_A_LAW;
774 case GST_BUFTYPE_MU_LAW:
775 alsa->format = SND_PCM_FORMAT_MU_LAW;
781 alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
782 alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
783 alsa->buffer_time = spec->buffer_time;
784 alsa->period_time = spec->latency_time;
785 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
797 gst_alsasrc_open (GstAudioSrc * asrc)
802 alsa = GST_ALSA_SRC (asrc);
804 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
805 SND_PCM_NONBLOCK), open_error);
808 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
816 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
817 (_("Could not open audio device for recording. "
818 "Device is being used by another application.")),
819 ("Device '%s' is busy", alsa->device));
821 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
822 (_("Could not open audio device for recording.")),
823 ("Recording open error on device '%s': %s", alsa->device,
824 snd_strerror (err)));
831 gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
836 alsa = GST_ALSA_SRC (asrc);
838 if (!alsasrc_parse_spec (alsa, spec))
841 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
843 CHECK (set_hwparams (alsa), hw_params_failed);
844 CHECK (set_swparams (alsa), sw_params_failed);
845 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
847 alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
848 spec->segsize = alsa->period_size * alsa->bpf;
849 spec->segtotal = alsa->buffer_size / alsa->period_size;
856 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
857 ("Error parsing spec"));
862 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
863 ("Could not set device to blocking: %s", snd_strerror (err)));
868 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
869 ("Setting of hwparams failed: %s", snd_strerror (err)));
874 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
875 ("Setting of swparams failed: %s", snd_strerror (err)));
880 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
881 ("Prepare failed: %s", snd_strerror (err)));
887 gst_alsasrc_unprepare (GstAudioSrc * asrc)
891 alsa = GST_ALSA_SRC (asrc);
893 snd_pcm_drop (alsa->handle);
894 snd_pcm_hw_free (alsa->handle);
895 snd_pcm_nonblock (alsa->handle, 1);
901 gst_alsasrc_close (GstAudioSrc * asrc)
903 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
905 snd_pcm_close (alsa->handle);
909 gst_alsa_mixer_free (alsa->mixer);
913 gst_caps_replace (&alsa->cached_caps, NULL);
919 * Underrun and suspend recovery
922 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
924 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
926 if (err == -EPIPE) { /* under-run */
927 err = snd_pcm_prepare (handle);
929 GST_WARNING_OBJECT (alsa,
930 "Can't recovery from underrun, prepare failed: %s",
933 } else if (err == -ESTRPIPE) {
934 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
935 g_usleep (100); /* wait until the suspend flag is released */
938 err = snd_pcm_prepare (handle);
940 GST_WARNING_OBJECT (alsa,
941 "Can't recovery from suspend, prepare failed: %s",
950 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
957 alsa = GST_ALSA_SRC (asrc);
959 cptr = length / alsa->bpf;
962 GST_ALSA_SRC_LOCK (asrc);
964 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
965 if (err == -EAGAIN) {
966 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
968 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
974 ptr += err * alsa->channels;
977 GST_ALSA_SRC_UNLOCK (asrc);
979 return length - (cptr * alsa->bpf);
983 GST_ALSA_SRC_UNLOCK (asrc);
984 return length; /* skip one period */
989 gst_alsasrc_delay (GstAudioSrc * asrc)
992 snd_pcm_sframes_t delay;
995 alsa = GST_ALSA_SRC (asrc);
997 res = snd_pcm_delay (alsa->handle, &delay);
998 if (G_UNLIKELY (res < 0)) {
999 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
1003 return CLAMP (delay, 0, alsa->buffer_size);
1007 gst_alsasrc_reset (GstAudioSrc * asrc)
1012 alsa = GST_ALSA_SRC (asrc);
1014 GST_ALSA_SRC_LOCK (asrc);
1015 GST_DEBUG_OBJECT (alsa, "drop");
1016 CHECK (snd_pcm_drop (alsa->handle), drop_error);
1017 GST_DEBUG_OBJECT (alsa, "prepare");
1018 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
1019 GST_DEBUG_OBJECT (alsa, "reset done");
1020 alsa->first_alsa_ts = GST_CLOCK_TIME_NONE;
1021 GST_ALSA_SRC_UNLOCK (asrc);
1028 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
1029 snd_strerror (err));
1030 GST_ALSA_SRC_UNLOCK (asrc);
1035 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
1036 snd_strerror (err));
1037 GST_ALSA_SRC_UNLOCK (asrc);