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"
51 #include "gst/glib-compat-private.h"
53 #include <gst/gst-i18n-plugin.h>
55 #define DEFAULT_PROP_DEVICE "default"
56 #define DEFAULT_PROP_DEVICE_NAME ""
57 #define DEFAULT_PROP_CARD_NAME ""
68 static void gst_alsasrc_init_interfaces (GType type);
70 GST_BOILERPLATE_FULL (GstAlsaSrc, gst_alsasrc, GstAudioSrc,
71 GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces);
73 GST_IMPLEMENT_ALSA_MIXER_METHODS (GstAlsaSrc, gst_alsasrc_mixer);
75 static void gst_alsasrc_finalize (GObject * object);
76 static void gst_alsasrc_set_property (GObject * object,
77 guint prop_id, const GValue * value, GParamSpec * pspec);
78 static void gst_alsasrc_get_property (GObject * object,
79 guint prop_id, GValue * value, GParamSpec * pspec);
81 static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc);
83 static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
84 static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
85 GstRingBufferSpec * spec);
86 static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);
87 static gboolean gst_alsasrc_close (GstAudioSrc * asrc);
88 static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length);
89 static guint gst_alsasrc_delay (GstAudioSrc * asrc);
90 static void gst_alsasrc_reset (GstAudioSrc * asrc);
92 /* AlsaSrc signals and args */
98 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
99 # define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
101 # define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
104 static GstStaticPadTemplate alsasrc_src_factory =
105 GST_STATIC_PAD_TEMPLATE ("src",
108 GST_STATIC_CAPS ("audio/x-raw-int, "
109 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
110 "signed = (boolean) { TRUE, FALSE }, "
113 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
115 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
116 "signed = (boolean) { TRUE, FALSE }, "
119 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
121 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
122 "signed = (boolean) { TRUE, FALSE }, "
125 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
127 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
128 "signed = (boolean) { TRUE, FALSE }, "
131 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
133 "signed = (boolean) { TRUE, FALSE }, "
136 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
140 gst_alsasrc_finalize (GObject * object)
142 GstAlsaSrc *src = GST_ALSA_SRC (object);
144 g_free (src->device);
145 g_mutex_free (src->alsa_lock);
147 G_OBJECT_CLASS (parent_class)->finalize (object);
151 gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type)
153 /* only support this one interface (wrapped by GstImplementsInterface) */
154 g_assert (interface_type == GST_TYPE_MIXER);
156 return gst_alsasrc_mixer_supported (this, interface_type);
160 gst_implements_interface_init (GstImplementsInterfaceClass * klass)
162 klass->supported = (gpointer) gst_alsasrc_interface_supported;
166 gst_alsasrc_init_interfaces (GType type)
168 static const GInterfaceInfo implements_iface_info = {
169 (GInterfaceInitFunc) gst_implements_interface_init,
173 static const GInterfaceInfo mixer_iface_info = {
174 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
179 g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
180 &implements_iface_info);
181 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
183 gst_alsa_type_add_device_property_probe_interface (type);
187 gst_alsasrc_base_init (gpointer g_class)
189 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
191 gst_element_class_set_details_simple (element_class,
192 "Audio source (ALSA)", "Source/Audio",
193 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
195 gst_element_class_add_static_pad_template (element_class,
196 &alsasrc_src_factory);
200 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
202 GObjectClass *gobject_class;
203 GstBaseSrcClass *gstbasesrc_class;
204 GstAudioSrcClass *gstaudiosrc_class;
206 gobject_class = (GObjectClass *) klass;
207 gstbasesrc_class = (GstBaseSrcClass *) klass;
208 gstaudiosrc_class = (GstAudioSrcClass *) klass;
210 gobject_class->finalize = gst_alsasrc_finalize;
211 gobject_class->get_property = gst_alsasrc_get_property;
212 gobject_class->set_property = gst_alsasrc_set_property;
214 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
216 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
217 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
218 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
219 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
220 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
221 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
222 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
224 g_object_class_install_property (gobject_class, PROP_DEVICE,
225 g_param_spec_string ("device", "Device",
226 "ALSA device, as defined in an asound configuration file",
227 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
229 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
230 g_param_spec_string ("device-name", "Device name",
231 "Human-readable name of the sound device",
232 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
234 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
235 g_param_spec_string ("card-name", "Card name",
236 "Human-readable name of the sound card",
237 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
241 gst_alsasrc_set_property (GObject * object, guint prop_id,
242 const GValue * value, GParamSpec * pspec)
246 src = GST_ALSA_SRC (object);
250 g_free (src->device);
251 src->device = g_value_dup_string (value);
252 if (src->device == NULL) {
253 src->device = g_strdup (DEFAULT_PROP_DEVICE);
257 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
263 gst_alsasrc_get_property (GObject * object, guint prop_id,
264 GValue * value, GParamSpec * pspec)
268 src = GST_ALSA_SRC (object);
272 g_value_set_string (value, src->device);
274 case PROP_DEVICE_NAME:
275 g_value_take_string (value,
276 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
277 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
280 g_value_take_string (value,
281 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
282 src->device, SND_PCM_STREAM_CAPTURE));
285 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
291 gst_alsasrc_init (GstAlsaSrc * alsasrc, GstAlsaSrcClass * g_class)
293 GST_DEBUG_OBJECT (alsasrc, "initializing");
295 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
296 alsasrc->cached_caps = NULL;
298 alsasrc->alsa_lock = g_mutex_new ();
301 #define CHECK(call, error) \
303 if ((err = call) < 0) \
309 gst_alsasrc_getcaps (GstBaseSrc * bsrc)
311 GstElementClass *element_class;
312 GstPadTemplate *pad_template;
316 src = GST_ALSA_SRC (bsrc);
318 if (src->handle == NULL) {
319 GST_DEBUG_OBJECT (src, "device not open, using template caps");
320 return NULL; /* base class will get template caps for us */
323 if (src->cached_caps) {
324 GST_LOG_OBJECT (src, "Returning cached caps");
325 return gst_caps_ref (src->cached_caps);
328 element_class = GST_ELEMENT_GET_CLASS (src);
329 pad_template = gst_element_class_get_pad_template (element_class, "src");
330 g_return_val_if_fail (pad_template != NULL, NULL);
332 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
333 gst_pad_template_get_caps (pad_template));
336 src->cached_caps = gst_caps_ref (caps);
339 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
345 set_hwparams (GstAlsaSrc * alsa)
349 snd_pcm_hw_params_t *params;
351 snd_pcm_hw_params_malloc (¶ms);
353 /* choose all parameters */
354 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
355 /* set the interleaved read/write format */
356 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
358 /* set the sample format */
359 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
361 /* set the count of channels */
362 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
364 /* set the stream rate */
366 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
368 if (rrate != alsa->rate)
371 if (alsa->buffer_time != -1) {
372 /* set the buffer time */
373 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
374 &alsa->buffer_time, NULL), buffer_time);
376 if (alsa->period_time != -1) {
377 /* set the period time */
378 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
379 &alsa->period_time, NULL), period_time);
382 /* write the parameters to device */
383 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
385 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
388 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
391 snd_pcm_hw_params_free (params);
397 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
398 ("Broken configuration for recording: no configurations available: %s",
399 snd_strerror (err)));
400 snd_pcm_hw_params_free (params);
405 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
406 ("Access type not available for recording: %s", snd_strerror (err)));
407 snd_pcm_hw_params_free (params);
412 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
413 ("Sample format not available for recording: %s", snd_strerror (err)));
414 snd_pcm_hw_params_free (params);
421 if ((alsa->channels) == 1)
422 msg = g_strdup (_("Could not open device for recording in mono mode."));
423 if ((alsa->channels) == 2)
424 msg = g_strdup (_("Could not open device for recording in stereo mode."));
425 if ((alsa->channels) > 2)
428 ("Could not open device for recording in %d-channel mode"),
430 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
431 ("%s", snd_strerror (err)));
433 snd_pcm_hw_params_free (params);
438 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
439 ("Rate %iHz not available for recording: %s",
440 alsa->rate, snd_strerror (err)));
441 snd_pcm_hw_params_free (params);
446 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
447 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
448 snd_pcm_hw_params_free (params);
453 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
454 ("Unable to set buffer time %i for recording: %s",
455 alsa->buffer_time, snd_strerror (err)));
456 snd_pcm_hw_params_free (params);
461 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
462 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
463 snd_pcm_hw_params_free (params);
468 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
469 ("Unable to set period time %i for recording: %s", alsa->period_time,
470 snd_strerror (err)));
471 snd_pcm_hw_params_free (params);
476 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
477 ("Unable to get period size for recording: %s", snd_strerror (err)));
478 snd_pcm_hw_params_free (params);
483 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
484 ("Unable to set hw params for recording: %s", snd_strerror (err)));
485 snd_pcm_hw_params_free (params);
491 set_swparams (GstAlsaSrc * alsa)
494 snd_pcm_sw_params_t *params;
496 snd_pcm_sw_params_malloc (¶ms);
498 /* get the current swparams */
499 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
500 /* allow the transfer when at least period_size samples can be processed */
501 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
502 alsa->period_size), set_avail);
503 /* start the transfer on first read */
504 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
505 0), start_threshold);
507 #if GST_CHECK_ALSA_VERSION(1,0,16)
508 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
510 /* align all transfers to 1 sample */
511 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
514 /* write the parameters to the recording device */
515 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
517 snd_pcm_sw_params_free (params);
523 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
524 ("Unable to determine current swparams for playback: %s",
525 snd_strerror (err)));
526 snd_pcm_sw_params_free (params);
531 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
532 ("Unable to set start threshold mode for playback: %s",
533 snd_strerror (err)));
534 snd_pcm_sw_params_free (params);
539 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
540 ("Unable to set avail min for playback: %s", snd_strerror (err)));
541 snd_pcm_sw_params_free (params);
544 #if !GST_CHECK_ALSA_VERSION(1,0,16)
547 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
548 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
549 snd_pcm_sw_params_free (params);
555 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
556 ("Unable to set sw params for playback: %s", snd_strerror (err)));
557 snd_pcm_sw_params_free (params);
563 alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec)
565 switch (spec->type) {
566 case GST_BUFTYPE_LINEAR:
567 alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
568 spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
570 case GST_BUFTYPE_FLOAT:
571 switch (spec->format) {
573 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
576 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
579 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
582 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
588 case GST_BUFTYPE_A_LAW:
589 alsa->format = SND_PCM_FORMAT_A_LAW;
591 case GST_BUFTYPE_MU_LAW:
592 alsa->format = SND_PCM_FORMAT_MU_LAW;
598 alsa->rate = spec->rate;
599 alsa->channels = spec->channels;
600 alsa->buffer_time = spec->buffer_time;
601 alsa->period_time = spec->latency_time;
602 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
614 gst_alsasrc_open (GstAudioSrc * asrc)
619 alsa = GST_ALSA_SRC (asrc);
621 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
622 SND_PCM_NONBLOCK), open_error);
625 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
633 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
634 (_("Could not open audio device for recording. "
635 "Device is being used by another application.")),
636 ("Device '%s' is busy", alsa->device));
638 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
639 (_("Could not open audio device for recording.")),
640 ("Recording open error on device '%s': %s", alsa->device,
641 snd_strerror (err)));
648 gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
653 alsa = GST_ALSA_SRC (asrc);
655 if (!alsasrc_parse_spec (alsa, spec))
658 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
660 CHECK (set_hwparams (alsa), hw_params_failed);
661 CHECK (set_swparams (alsa), sw_params_failed);
662 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
664 alsa->bytes_per_sample = spec->bytes_per_sample;
665 spec->segsize = alsa->period_size * spec->bytes_per_sample;
666 spec->segtotal = alsa->buffer_size / alsa->period_size;
667 spec->silence_sample[0] = 0;
668 spec->silence_sample[1] = 0;
669 spec->silence_sample[2] = 0;
670 spec->silence_sample[3] = 0;
677 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
678 ("Error parsing spec"));
683 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
684 ("Could not set device to blocking: %s", snd_strerror (err)));
689 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
690 ("Setting of hwparams failed: %s", snd_strerror (err)));
695 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
696 ("Setting of swparams failed: %s", snd_strerror (err)));
701 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
702 ("Prepare failed: %s", snd_strerror (err)));
708 gst_alsasrc_unprepare (GstAudioSrc * asrc)
712 alsa = GST_ALSA_SRC (asrc);
714 snd_pcm_drop (alsa->handle);
715 snd_pcm_hw_free (alsa->handle);
716 snd_pcm_nonblock (alsa->handle, 1);
722 gst_alsasrc_close (GstAudioSrc * asrc)
724 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
726 snd_pcm_close (alsa->handle);
730 gst_alsa_mixer_free (alsa->mixer);
734 gst_caps_replace (&alsa->cached_caps, NULL);
740 * Underrun and suspend recovery
743 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
745 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
747 if (err == -EPIPE) { /* under-run */
748 err = snd_pcm_prepare (handle);
750 GST_WARNING_OBJECT (alsa,
751 "Can't recovery from underrun, prepare failed: %s",
754 } else if (err == -ESTRPIPE) {
755 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
756 g_usleep (100); /* wait until the suspend flag is released */
759 err = snd_pcm_prepare (handle);
761 GST_WARNING_OBJECT (alsa,
762 "Can't recovery from suspend, prepare failed: %s",
771 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
778 alsa = GST_ALSA_SRC (asrc);
780 cptr = length / alsa->bytes_per_sample;
783 GST_ALSA_SRC_LOCK (asrc);
785 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
786 if (err == -EAGAIN) {
787 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
789 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
795 ptr += err * alsa->channels;
798 GST_ALSA_SRC_UNLOCK (asrc);
800 return length - (cptr * alsa->bytes_per_sample);
804 GST_ALSA_SRC_UNLOCK (asrc);
805 return length; /* skip one period */
810 gst_alsasrc_delay (GstAudioSrc * asrc)
813 snd_pcm_sframes_t delay;
816 alsa = GST_ALSA_SRC (asrc);
818 res = snd_pcm_delay (alsa->handle, &delay);
819 if (G_UNLIKELY (res < 0)) {
820 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
824 return CLAMP (delay, 0, alsa->buffer_size);
828 gst_alsasrc_reset (GstAudioSrc * asrc)
833 alsa = GST_ALSA_SRC (asrc);
835 GST_ALSA_SRC_LOCK (asrc);
836 GST_DEBUG_OBJECT (alsa, "drop");
837 CHECK (snd_pcm_drop (alsa->handle), drop_error);
838 GST_DEBUG_OBJECT (alsa, "prepare");
839 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
840 GST_DEBUG_OBJECT (alsa, "reset done");
841 GST_ALSA_SRC_UNLOCK (asrc);
848 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
850 GST_ALSA_SRC_UNLOCK (asrc);
855 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
857 GST_ALSA_SRC_UNLOCK (asrc);