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);
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);
91 /* AlsaSrc signals and args */
97 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
98 # define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
100 # define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
103 static GstStaticPadTemplate alsasrc_src_factory =
104 GST_STATIC_PAD_TEMPLATE ("src",
107 GST_STATIC_CAPS ("audio/x-raw-int, "
108 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
109 "signed = (boolean) { TRUE, FALSE }, "
112 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
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 "signed = (boolean) { TRUE, FALSE }, "
135 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
139 gst_alsasrc_finalize (GObject * object)
141 GstAlsaSrc *src = GST_ALSA_SRC (object);
143 g_free (src->device);
144 g_mutex_free (src->alsa_lock);
146 G_OBJECT_CLASS (parent_class)->finalize (object);
150 gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type)
152 /* only support this one interface (wrapped by GstImplementsInterface) */
153 g_assert (interface_type == GST_TYPE_MIXER);
155 return gst_alsasrc_mixer_supported (this, interface_type);
159 gst_implements_interface_init (GstImplementsInterfaceClass * klass)
161 klass->supported = (gpointer) gst_alsasrc_interface_supported;
165 gst_alsasrc_init_interfaces (GType type)
167 static const GInterfaceInfo implements_iface_info = {
168 (GInterfaceInitFunc) gst_implements_interface_init,
172 static const GInterfaceInfo mixer_iface_info = {
173 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
178 g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
179 &implements_iface_info);
180 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
182 gst_alsa_type_add_device_property_probe_interface (type);
186 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
188 GObjectClass *gobject_class;
189 GstElementClass *gstelement_class;
190 GstBaseSrcClass *gstbasesrc_class;
191 GstAudioSrcClass *gstaudiosrc_class;
193 gobject_class = (GObjectClass *) klass;
194 gstelement_class = (GstElementClass *) klass;
195 gstbasesrc_class = (GstBaseSrcClass *) klass;
196 gstaudiosrc_class = (GstAudioSrcClass *) klass;
198 gobject_class->finalize = gst_alsasrc_finalize;
199 gobject_class->get_property = gst_alsasrc_get_property;
200 gobject_class->set_property = gst_alsasrc_set_property;
202 gst_element_class_set_details_simple (gstelement_class,
203 "Audio source (ALSA)", "Source/Audio",
204 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
206 gst_element_class_add_pad_template (gstelement_class,
207 gst_static_pad_template_get (&alsasrc_src_factory));
209 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
211 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
212 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
213 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
214 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
215 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
216 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
217 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
219 g_object_class_install_property (gobject_class, PROP_DEVICE,
220 g_param_spec_string ("device", "Device",
221 "ALSA device, as defined in an asound configuration file",
222 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
224 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
225 g_param_spec_string ("device-name", "Device name",
226 "Human-readable name of the sound device",
227 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
229 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
230 g_param_spec_string ("card-name", "Card name",
231 "Human-readable name of the sound card",
232 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
236 gst_alsasrc_set_property (GObject * object, guint prop_id,
237 const GValue * value, GParamSpec * pspec)
241 src = GST_ALSA_SRC (object);
245 g_free (src->device);
246 src->device = g_value_dup_string (value);
247 if (src->device == NULL) {
248 src->device = g_strdup (DEFAULT_PROP_DEVICE);
252 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
258 gst_alsasrc_get_property (GObject * object, guint prop_id,
259 GValue * value, GParamSpec * pspec)
263 src = GST_ALSA_SRC (object);
267 g_value_set_string (value, src->device);
269 case PROP_DEVICE_NAME:
270 g_value_take_string (value,
271 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
272 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
275 g_value_take_string (value,
276 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
277 src->device, SND_PCM_STREAM_CAPTURE));
280 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
286 gst_alsasrc_init (GstAlsaSrc * alsasrc)
288 GST_DEBUG_OBJECT (alsasrc, "initializing");
290 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
291 alsasrc->cached_caps = NULL;
293 alsasrc->alsa_lock = g_mutex_new ();
296 #define CHECK(call, error) \
298 if ((err = call) < 0) \
304 gst_alsasrc_getcaps (GstBaseSrc * bsrc)
306 GstElementClass *element_class;
307 GstPadTemplate *pad_template;
311 src = GST_ALSA_SRC (bsrc);
313 if (src->handle == NULL) {
314 GST_DEBUG_OBJECT (src, "device not open, using template caps");
315 return NULL; /* base class will get template caps for us */
318 if (src->cached_caps) {
319 GST_LOG_OBJECT (src, "Returning cached caps");
320 return gst_caps_ref (src->cached_caps);
323 element_class = GST_ELEMENT_GET_CLASS (src);
324 pad_template = gst_element_class_get_pad_template (element_class, "src");
325 g_return_val_if_fail (pad_template != NULL, NULL);
327 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
328 gst_pad_template_get_caps (pad_template));
331 src->cached_caps = gst_caps_ref (caps);
334 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
340 set_hwparams (GstAlsaSrc * alsa)
344 snd_pcm_hw_params_t *params;
346 snd_pcm_hw_params_malloc (¶ms);
348 /* choose all parameters */
349 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
350 /* set the interleaved read/write format */
351 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
353 /* set the sample format */
354 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
356 /* set the count of channels */
357 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
359 /* set the stream rate */
361 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
363 if (rrate != alsa->rate)
366 if (alsa->buffer_time != -1) {
367 /* set the buffer time */
368 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
369 &alsa->buffer_time, NULL), buffer_time);
371 if (alsa->period_time != -1) {
372 /* set the period time */
373 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
374 &alsa->period_time, NULL), period_time);
377 /* write the parameters to device */
378 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
380 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
383 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
386 snd_pcm_hw_params_free (params);
392 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
393 ("Broken configuration for recording: no configurations available: %s",
394 snd_strerror (err)));
395 snd_pcm_hw_params_free (params);
400 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
401 ("Access type not available for recording: %s", snd_strerror (err)));
402 snd_pcm_hw_params_free (params);
407 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
408 ("Sample format not available for recording: %s", snd_strerror (err)));
409 snd_pcm_hw_params_free (params);
416 if ((alsa->channels) == 1)
417 msg = g_strdup (_("Could not open device for recording in mono mode."));
418 if ((alsa->channels) == 2)
419 msg = g_strdup (_("Could not open device for recording in stereo mode."));
420 if ((alsa->channels) > 2)
423 ("Could not open device for recording in %d-channel mode"),
425 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
426 ("%s", snd_strerror (err)));
428 snd_pcm_hw_params_free (params);
433 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
434 ("Rate %iHz not available for recording: %s",
435 alsa->rate, snd_strerror (err)));
436 snd_pcm_hw_params_free (params);
441 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
442 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
443 snd_pcm_hw_params_free (params);
448 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
449 ("Unable to set buffer time %i for recording: %s",
450 alsa->buffer_time, snd_strerror (err)));
451 snd_pcm_hw_params_free (params);
456 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
457 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
458 snd_pcm_hw_params_free (params);
463 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
464 ("Unable to set period time %i for recording: %s", alsa->period_time,
465 snd_strerror (err)));
466 snd_pcm_hw_params_free (params);
471 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
472 ("Unable to get period size for recording: %s", snd_strerror (err)));
473 snd_pcm_hw_params_free (params);
478 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
479 ("Unable to set hw params for recording: %s", snd_strerror (err)));
480 snd_pcm_hw_params_free (params);
486 set_swparams (GstAlsaSrc * alsa)
489 snd_pcm_sw_params_t *params;
491 snd_pcm_sw_params_malloc (¶ms);
493 /* get the current swparams */
494 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
495 /* allow the transfer when at least period_size samples can be processed */
496 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
497 alsa->period_size), set_avail);
498 /* start the transfer on first read */
499 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
500 0), start_threshold);
502 #if GST_CHECK_ALSA_VERSION(1,0,16)
503 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
505 /* align all transfers to 1 sample */
506 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
509 /* write the parameters to the recording device */
510 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
512 snd_pcm_sw_params_free (params);
518 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
519 ("Unable to determine current swparams for playback: %s",
520 snd_strerror (err)));
521 snd_pcm_sw_params_free (params);
526 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
527 ("Unable to set start threshold mode for playback: %s",
528 snd_strerror (err)));
529 snd_pcm_sw_params_free (params);
534 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
535 ("Unable to set avail min for playback: %s", snd_strerror (err)));
536 snd_pcm_sw_params_free (params);
539 #if !GST_CHECK_ALSA_VERSION(1,0,16)
542 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
543 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
544 snd_pcm_sw_params_free (params);
550 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
551 ("Unable to set sw params for playback: %s", snd_strerror (err)));
552 snd_pcm_sw_params_free (params);
558 alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec)
560 switch (spec->type) {
561 case GST_BUFTYPE_LINEAR:
562 alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
563 spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
565 case GST_BUFTYPE_FLOAT:
566 switch (spec->format) {
568 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
571 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
574 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
577 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
583 case GST_BUFTYPE_A_LAW:
584 alsa->format = SND_PCM_FORMAT_A_LAW;
586 case GST_BUFTYPE_MU_LAW:
587 alsa->format = SND_PCM_FORMAT_MU_LAW;
593 alsa->rate = spec->rate;
594 alsa->channels = spec->channels;
595 alsa->buffer_time = spec->buffer_time;
596 alsa->period_time = spec->latency_time;
597 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
609 gst_alsasrc_open (GstAudioSrc * asrc)
614 alsa = GST_ALSA_SRC (asrc);
616 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
617 SND_PCM_NONBLOCK), open_error);
620 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
628 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
629 (_("Could not open audio device for recording. "
630 "Device is being used by another application.")),
631 ("Device '%s' is busy", alsa->device));
633 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
634 (_("Could not open audio device for recording.")),
635 ("Recording open error on device '%s': %s", alsa->device,
636 snd_strerror (err)));
643 gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
648 alsa = GST_ALSA_SRC (asrc);
650 if (!alsasrc_parse_spec (alsa, spec))
653 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
655 CHECK (set_hwparams (alsa), hw_params_failed);
656 CHECK (set_swparams (alsa), sw_params_failed);
657 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
659 alsa->bytes_per_sample = spec->bytes_per_sample;
660 spec->segsize = alsa->period_size * spec->bytes_per_sample;
661 spec->segtotal = alsa->buffer_size / alsa->period_size;
662 spec->silence_sample[0] = 0;
663 spec->silence_sample[1] = 0;
664 spec->silence_sample[2] = 0;
665 spec->silence_sample[3] = 0;
672 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
673 ("Error parsing spec"));
678 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
679 ("Could not set device to blocking: %s", snd_strerror (err)));
684 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
685 ("Setting of hwparams failed: %s", snd_strerror (err)));
690 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
691 ("Setting of swparams failed: %s", snd_strerror (err)));
696 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
697 ("Prepare failed: %s", snd_strerror (err)));
703 gst_alsasrc_unprepare (GstAudioSrc * asrc)
707 alsa = GST_ALSA_SRC (asrc);
709 snd_pcm_drop (alsa->handle);
710 snd_pcm_hw_free (alsa->handle);
711 snd_pcm_nonblock (alsa->handle, 1);
717 gst_alsasrc_close (GstAudioSrc * asrc)
719 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
721 snd_pcm_close (alsa->handle);
725 gst_alsa_mixer_free (alsa->mixer);
729 gst_caps_replace (&alsa->cached_caps, NULL);
735 * Underrun and suspend recovery
738 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
740 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
742 if (err == -EPIPE) { /* under-run */
743 err = snd_pcm_prepare (handle);
745 GST_WARNING_OBJECT (alsa,
746 "Can't recovery from underrun, prepare failed: %s",
749 } else if (err == -ESTRPIPE) {
750 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
751 g_usleep (100); /* wait until the suspend flag is released */
754 err = snd_pcm_prepare (handle);
756 GST_WARNING_OBJECT (alsa,
757 "Can't recovery from suspend, prepare failed: %s",
766 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
773 alsa = GST_ALSA_SRC (asrc);
775 cptr = length / alsa->bytes_per_sample;
778 GST_ALSA_SRC_LOCK (asrc);
780 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
781 if (err == -EAGAIN) {
782 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
784 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
790 ptr += err * alsa->channels;
793 GST_ALSA_SRC_UNLOCK (asrc);
795 return length - (cptr * alsa->bytes_per_sample);
799 GST_ALSA_SRC_UNLOCK (asrc);
800 return length; /* skip one period */
805 gst_alsasrc_delay (GstAudioSrc * asrc)
808 snd_pcm_sframes_t delay;
811 alsa = GST_ALSA_SRC (asrc);
813 res = snd_pcm_delay (alsa->handle, &delay);
814 if (G_UNLIKELY (res < 0)) {
815 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
819 return CLAMP (delay, 0, alsa->buffer_size);
823 gst_alsasrc_reset (GstAudioSrc * asrc)
828 alsa = GST_ALSA_SRC (asrc);
830 GST_ALSA_SRC_LOCK (asrc);
831 GST_DEBUG_OBJECT (alsa, "drop");
832 CHECK (snd_pcm_drop (alsa->handle), drop_error);
833 GST_DEBUG_OBJECT (alsa, "prepare");
834 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
835 GST_DEBUG_OBJECT (alsa, "reset done");
836 GST_ALSA_SRC_UNLOCK (asrc);
843 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
845 GST_ALSA_SRC_UNLOCK (asrc);
850 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
852 GST_ALSA_SRC_UNLOCK (asrc);