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 GstRingBufferSpec * 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-int, "
114 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
115 "signed = (boolean) { TRUE, FALSE }, "
118 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
120 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
121 "signed = (boolean) { TRUE, FALSE }, "
124 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
126 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
127 "signed = (boolean) { TRUE, FALSE }, "
130 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
132 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
133 "signed = (boolean) { TRUE, FALSE }, "
136 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
138 "signed = (boolean) { TRUE, FALSE }, "
141 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
145 gst_alsasrc_finalize (GObject * object)
147 GstAlsaSrc *src = GST_ALSA_SRC (object);
149 g_free (src->device);
150 g_mutex_free (src->alsa_lock);
152 G_OBJECT_CLASS (parent_class)->finalize (object);
156 gst_alsasrc_init_interfaces (GType type)
158 static const GInterfaceInfo mixer_iface_info = {
159 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
164 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
166 gst_alsa_type_add_device_property_probe_interface (type);
170 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
172 GObjectClass *gobject_class;
173 GstElementClass *gstelement_class;
174 GstBaseSrcClass *gstbasesrc_class;
175 GstAudioSrcClass *gstaudiosrc_class;
177 gobject_class = (GObjectClass *) klass;
178 gstelement_class = (GstElementClass *) klass;
179 gstbasesrc_class = (GstBaseSrcClass *) klass;
180 gstaudiosrc_class = (GstAudioSrcClass *) klass;
182 gobject_class->finalize = gst_alsasrc_finalize;
183 gobject_class->get_property = gst_alsasrc_get_property;
184 gobject_class->set_property = gst_alsasrc_set_property;
186 gst_element_class_set_details_simple (gstelement_class,
187 "Audio source (ALSA)", "Source/Audio",
188 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
190 gst_element_class_add_pad_template (gstelement_class,
191 gst_static_pad_template_get (&alsasrc_src_factory));
193 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
195 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
196 gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_alsasrc_create);
198 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
199 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
200 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
201 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
202 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
203 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
204 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
206 g_object_class_install_property (gobject_class, PROP_DEVICE,
207 g_param_spec_string ("device", "Device",
208 "ALSA device, as defined in an asound configuration file",
209 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
211 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
212 g_param_spec_string ("device-name", "Device name",
213 "Human-readable name of the sound device",
214 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
216 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
217 g_param_spec_string ("card-name", "Card name",
218 "Human-readable name of the sound card",
219 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
223 gst_alsasrc_get_timestamp (GstAlsaSrc * src)
225 snd_pcm_status_t *status;
226 snd_htimestamp_t tstamp;
227 GstClockTime timestamp;
228 snd_pcm_uframes_t availmax;
230 GST_DEBUG_OBJECT (src, "Getting alsa timestamp!");
233 GST_ERROR_OBJECT (src, "No alsa handle created yet !");
237 if (snd_pcm_status_malloc (&status) != 0) {
238 GST_ERROR_OBJECT (src, "snd_pcm_status_malloc failed");
241 if (snd_pcm_status (src->handle, status) != 0) {
242 GST_ERROR_OBJECT (src, "snd_pcm_status failed");
245 /* get high resolution time stamp from driver */
246 snd_pcm_status_get_htstamp (status, &tstamp);
247 timestamp = GST_TIMESPEC_TO_TIME (tstamp);
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 timestamp -= gst_util_uint64_scale_int (availmax * 2, GST_SECOND, src->rate);
254 /* Compensate for the delay until the package is available */
255 timestamp += gst_util_uint64_scale_int (snd_pcm_status_get_delay (status),
256 GST_SECOND, src->rate);
258 snd_pcm_status_free (status);
260 GST_DEBUG_OBJECT (src, "ALSA timestamp : %" GST_TIME_FORMAT,
261 GST_TIME_ARGS (timestamp));
266 gst_alsasrc_set_property (GObject * object, guint prop_id,
267 const GValue * value, GParamSpec * pspec)
271 src = GST_ALSA_SRC (object);
275 g_free (src->device);
276 src->device = g_value_dup_string (value);
277 if (src->device == NULL) {
278 src->device = g_strdup (DEFAULT_PROP_DEVICE);
282 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
288 gst_alsasrc_get_property (GObject * object, guint prop_id,
289 GValue * value, GParamSpec * pspec)
293 src = GST_ALSA_SRC (object);
297 g_value_set_string (value, src->device);
299 case PROP_DEVICE_NAME:
300 g_value_take_string (value,
301 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
302 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
305 g_value_take_string (value,
306 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
307 src->device, SND_PCM_STREAM_CAPTURE));
310 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
315 static GstStateChangeReturn
316 gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
318 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
319 GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
320 GstAlsaSrc *asrc = GST_ALSA_SRC (element);
323 switch (transition) {
324 /* Show the compiler that we care */
325 case GST_STATE_CHANGE_NULL_TO_READY:
326 case GST_STATE_CHANGE_READY_TO_PAUSED:
327 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
328 case GST_STATE_CHANGE_PAUSED_TO_READY:
329 case GST_STATE_CHANGE_READY_TO_NULL:
332 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
334 asrc->driver_timestamps = FALSE;
335 if (GST_IS_SYSTEM_CLOCK (clk)) {
337 g_object_get (clk, "clock-type", &clocktype, NULL);
338 if (clocktype == GST_CLOCK_TYPE_MONOTONIC) {
339 asrc->driver_timestamps = TRUE;
344 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
350 gst_alsasrc_create (GstBaseSrc * bsrc, guint64 offset, guint length,
353 GstFlowReturn ret = GST_FLOW_OK;
354 GstAlsaSrc *asrc = GST_ALSA_SRC (bsrc);
357 GST_BASE_SRC_CLASS (parent_class)->create (bsrc, offset, length, outbuf);
358 if (asrc->driver_timestamps == TRUE && *outbuf) {
359 GST_BUFFER_TIMESTAMP (*outbuf) =
360 gst_alsasrc_get_timestamp ((GstAlsaSrc *) bsrc);
367 gst_alsasrc_init (GstAlsaSrc * alsasrc)
369 GST_DEBUG_OBJECT (alsasrc, "initializing");
371 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
372 alsasrc->cached_caps = NULL;
373 alsasrc->driver_timestamps = FALSE;
375 alsasrc->alsa_lock = g_mutex_new ();
378 #define CHECK(call, error) \
380 if ((err = call) < 0) \
386 gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
388 GstElementClass *element_class;
389 GstPadTemplate *pad_template;
391 GstCaps *caps, *templ_caps;
393 src = GST_ALSA_SRC (bsrc);
395 if (src->handle == NULL) {
396 GST_DEBUG_OBJECT (src, "device not open, using template caps");
397 return NULL; /* base class will get template caps for us */
400 if (src->cached_caps) {
401 GST_LOG_OBJECT (src, "Returning cached caps");
403 return gst_caps_intersect_full (filter, src->cached_caps,
404 GST_CAPS_INTERSECT_FIRST);
406 return gst_caps_ref (src->cached_caps);
409 element_class = GST_ELEMENT_GET_CLASS (src);
410 pad_template = gst_element_class_get_pad_template (element_class, "src");
411 g_return_val_if_fail (pad_template != NULL, NULL);
413 templ_caps = gst_pad_template_get_caps (pad_template);
414 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
416 gst_caps_unref (templ_caps);
419 src->cached_caps = gst_caps_ref (caps);
422 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
425 GstCaps *intersection;
428 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
429 gst_caps_unref (caps);
437 set_hwparams (GstAlsaSrc * alsa)
441 snd_pcm_hw_params_t *params;
443 snd_pcm_hw_params_malloc (¶ms);
445 /* choose all parameters */
446 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
447 /* set the interleaved read/write format */
448 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
450 /* set the sample format */
451 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
453 /* set the count of channels */
454 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
456 /* set the stream rate */
458 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
460 if (rrate != alsa->rate)
463 if (alsa->buffer_time != -1) {
464 /* set the buffer time */
465 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
466 &alsa->buffer_time, NULL), buffer_time);
468 if (alsa->period_time != -1) {
469 /* set the period time */
470 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
471 &alsa->period_time, NULL), period_time);
474 /* write the parameters to device */
475 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
477 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
480 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
483 snd_pcm_hw_params_free (params);
489 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
490 ("Broken configuration for recording: no configurations available: %s",
491 snd_strerror (err)));
492 snd_pcm_hw_params_free (params);
497 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
498 ("Access type not available for recording: %s", snd_strerror (err)));
499 snd_pcm_hw_params_free (params);
504 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
505 ("Sample format not available for recording: %s", snd_strerror (err)));
506 snd_pcm_hw_params_free (params);
513 if ((alsa->channels) == 1)
514 msg = g_strdup (_("Could not open device for recording in mono mode."));
515 if ((alsa->channels) == 2)
516 msg = g_strdup (_("Could not open device for recording in stereo mode."));
517 if ((alsa->channels) > 2)
520 ("Could not open device for recording in %d-channel mode"),
522 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
523 ("%s", snd_strerror (err)));
525 snd_pcm_hw_params_free (params);
530 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
531 ("Rate %iHz not available for recording: %s",
532 alsa->rate, snd_strerror (err)));
533 snd_pcm_hw_params_free (params);
538 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
539 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
540 snd_pcm_hw_params_free (params);
545 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
546 ("Unable to set buffer time %i for recording: %s",
547 alsa->buffer_time, snd_strerror (err)));
548 snd_pcm_hw_params_free (params);
553 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
554 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
555 snd_pcm_hw_params_free (params);
560 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
561 ("Unable to set period time %i for recording: %s", alsa->period_time,
562 snd_strerror (err)));
563 snd_pcm_hw_params_free (params);
568 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
569 ("Unable to get period size for recording: %s", snd_strerror (err)));
570 snd_pcm_hw_params_free (params);
575 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
576 ("Unable to set hw params for recording: %s", snd_strerror (err)));
577 snd_pcm_hw_params_free (params);
583 set_swparams (GstAlsaSrc * alsa)
586 snd_pcm_sw_params_t *params;
588 snd_pcm_sw_params_malloc (¶ms);
590 /* get the current swparams */
591 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
592 /* allow the transfer when at least period_size samples can be processed */
593 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
594 alsa->period_size), set_avail);
595 /* start the transfer on first read */
596 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
597 0), start_threshold);
599 #if GST_CHECK_ALSA_VERSION(1,0,16)
600 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
602 /* align all transfers to 1 sample */
603 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
606 /* write the parameters to the recording device */
607 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
609 snd_pcm_sw_params_free (params);
615 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
616 ("Unable to determine current swparams for playback: %s",
617 snd_strerror (err)));
618 snd_pcm_sw_params_free (params);
623 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
624 ("Unable to set start threshold mode for playback: %s",
625 snd_strerror (err)));
626 snd_pcm_sw_params_free (params);
631 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
632 ("Unable to set avail min for playback: %s", snd_strerror (err)));
633 snd_pcm_sw_params_free (params);
636 #if !GST_CHECK_ALSA_VERSION(1,0,16)
639 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
640 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
641 snd_pcm_sw_params_free (params);
647 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
648 ("Unable to set sw params for playback: %s", snd_strerror (err)));
649 snd_pcm_sw_params_free (params);
655 alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec)
657 switch (spec->type) {
658 case GST_BUFTYPE_LINEAR:
659 alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
660 spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
662 case GST_BUFTYPE_FLOAT:
663 switch (spec->format) {
665 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
668 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
671 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
674 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
680 case GST_BUFTYPE_A_LAW:
681 alsa->format = SND_PCM_FORMAT_A_LAW;
683 case GST_BUFTYPE_MU_LAW:
684 alsa->format = SND_PCM_FORMAT_MU_LAW;
690 alsa->rate = spec->rate;
691 alsa->channels = spec->channels;
692 alsa->buffer_time = spec->buffer_time;
693 alsa->period_time = spec->latency_time;
694 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
706 gst_alsasrc_open (GstAudioSrc * asrc)
711 alsa = GST_ALSA_SRC (asrc);
713 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
714 SND_PCM_NONBLOCK), open_error);
717 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
725 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
726 (_("Could not open audio device for recording. "
727 "Device is being used by another application.")),
728 ("Device '%s' is busy", alsa->device));
730 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
731 (_("Could not open audio device for recording.")),
732 ("Recording open error on device '%s': %s", alsa->device,
733 snd_strerror (err)));
740 gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
745 alsa = GST_ALSA_SRC (asrc);
747 if (!alsasrc_parse_spec (alsa, spec))
750 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
752 CHECK (set_hwparams (alsa), hw_params_failed);
753 CHECK (set_swparams (alsa), sw_params_failed);
754 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
756 alsa->bytes_per_sample = spec->bytes_per_sample;
757 spec->segsize = alsa->period_size * spec->bytes_per_sample;
758 spec->segtotal = alsa->buffer_size / alsa->period_size;
759 spec->silence_sample[0] = 0;
760 spec->silence_sample[1] = 0;
761 spec->silence_sample[2] = 0;
762 spec->silence_sample[3] = 0;
769 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
770 ("Error parsing spec"));
775 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
776 ("Could not set device to blocking: %s", snd_strerror (err)));
781 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
782 ("Setting of hwparams failed: %s", snd_strerror (err)));
787 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
788 ("Setting of swparams failed: %s", snd_strerror (err)));
793 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
794 ("Prepare failed: %s", snd_strerror (err)));
800 gst_alsasrc_unprepare (GstAudioSrc * asrc)
804 alsa = GST_ALSA_SRC (asrc);
806 snd_pcm_drop (alsa->handle);
807 snd_pcm_hw_free (alsa->handle);
808 snd_pcm_nonblock (alsa->handle, 1);
814 gst_alsasrc_close (GstAudioSrc * asrc)
816 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
818 snd_pcm_close (alsa->handle);
822 gst_alsa_mixer_free (alsa->mixer);
826 gst_caps_replace (&alsa->cached_caps, NULL);
832 * Underrun and suspend recovery
835 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
837 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
839 if (err == -EPIPE) { /* under-run */
840 err = snd_pcm_prepare (handle);
842 GST_WARNING_OBJECT (alsa,
843 "Can't recovery from underrun, prepare failed: %s",
846 } else if (err == -ESTRPIPE) {
847 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
848 g_usleep (100); /* wait until the suspend flag is released */
851 err = snd_pcm_prepare (handle);
853 GST_WARNING_OBJECT (alsa,
854 "Can't recovery from suspend, prepare failed: %s",
863 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
870 alsa = GST_ALSA_SRC (asrc);
872 cptr = length / alsa->bytes_per_sample;
875 GST_ALSA_SRC_LOCK (asrc);
877 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
878 if (err == -EAGAIN) {
879 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
881 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
887 ptr += err * alsa->channels;
890 GST_ALSA_SRC_UNLOCK (asrc);
892 return length - (cptr * alsa->bytes_per_sample);
896 GST_ALSA_SRC_UNLOCK (asrc);
897 return length; /* skip one period */
902 gst_alsasrc_delay (GstAudioSrc * asrc)
905 snd_pcm_sframes_t delay;
908 alsa = GST_ALSA_SRC (asrc);
910 res = snd_pcm_delay (alsa->handle, &delay);
911 if (G_UNLIKELY (res < 0)) {
912 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
916 return CLAMP (delay, 0, alsa->buffer_size);
920 gst_alsasrc_reset (GstAudioSrc * asrc)
925 alsa = GST_ALSA_SRC (asrc);
927 GST_ALSA_SRC_LOCK (asrc);
928 GST_DEBUG_OBJECT (alsa, "drop");
929 CHECK (snd_pcm_drop (alsa->handle), drop_error);
930 GST_DEBUG_OBJECT (alsa, "prepare");
931 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
932 GST_DEBUG_OBJECT (alsa, "reset done");
933 GST_ALSA_SRC_UNLOCK (asrc);
940 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
942 GST_ALSA_SRC_UNLOCK (asrc);
947 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
949 GST_ALSA_SRC_UNLOCK (asrc);