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 /* elementfactory information */
55 static const GstElementDetails gst_alsasrc_details =
56 GST_ELEMENT_DETAILS ("Audio source (ALSA)",
58 "Read from a sound card via ALSA",
59 "Wim Taymans <wim@fluendo.com>");
61 #define DEFAULT_PROP_DEVICE "default"
62 #define DEFAULT_PROP_DEVICE_NAME ""
71 static void gst_alsasrc_init_interfaces (GType type);
73 GST_BOILERPLATE_FULL (GstAlsaSrc, gst_alsasrc, GstAudioSrc,
74 GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces);
76 GST_IMPLEMENT_ALSA_MIXER_METHODS (GstAlsaSrc, gst_alsasrc_mixer);
78 static void gst_alsasrc_finalize (GObject * object);
79 static void gst_alsasrc_set_property (GObject * object,
80 guint prop_id, const GValue * value, GParamSpec * pspec);
81 static void gst_alsasrc_get_property (GObject * object,
82 guint prop_id, GValue * value, GParamSpec * pspec);
84 static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc);
86 static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
87 static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
88 GstRingBufferSpec * spec);
89 static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);
90 static gboolean gst_alsasrc_close (GstAudioSrc * asrc);
91 static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length);
92 static guint gst_alsasrc_delay (GstAudioSrc * asrc);
93 static void gst_alsasrc_reset (GstAudioSrc * asrc);
95 /* AlsaSrc signals and args */
101 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
102 # define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
104 # define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
107 static GstStaticPadTemplate alsasrc_src_factory =
108 GST_STATIC_PAD_TEMPLATE ("src",
111 GST_STATIC_CAPS ("audio/x-raw-int, "
112 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
113 "signed = (boolean) { TRUE, FALSE }, "
116 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
118 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
119 "signed = (boolean) { TRUE, FALSE }, "
122 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
124 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
125 "signed = (boolean) { TRUE, FALSE }, "
128 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
130 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
131 "signed = (boolean) { TRUE, FALSE }, "
134 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
136 "signed = (boolean) { TRUE, FALSE }, "
139 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
143 gst_alsasrc_finalize (GObject * object)
145 GstAlsaSrc *src = GST_ALSA_SRC (object);
147 g_free (src->device);
148 g_mutex_free (src->alsa_lock);
150 G_OBJECT_CLASS (parent_class)->finalize (object);
154 gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type)
156 /* only support this one interface (wrapped by GstImplementsInterface) */
157 g_assert (interface_type == GST_TYPE_MIXER);
159 return gst_alsasrc_mixer_supported (this, interface_type);
163 gst_implements_interface_init (GstImplementsInterfaceClass * klass)
165 klass->supported = (gpointer) gst_alsasrc_interface_supported;
169 gst_alsasrc_init_interfaces (GType type)
171 static const GInterfaceInfo implements_iface_info = {
172 (GInterfaceInitFunc) gst_implements_interface_init,
176 static const GInterfaceInfo mixer_iface_info = {
177 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
182 g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
183 &implements_iface_info);
184 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
186 gst_alsa_type_add_device_property_probe_interface (type);
190 gst_alsasrc_base_init (gpointer g_class)
192 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
194 gst_element_class_set_details (element_class, &gst_alsasrc_details);
196 gst_element_class_add_pad_template (element_class,
197 gst_static_pad_template_get (&alsasrc_src_factory));
201 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
203 GObjectClass *gobject_class;
204 GstBaseSrcClass *gstbasesrc_class;
205 GstAudioSrcClass *gstaudiosrc_class;
207 gobject_class = (GObjectClass *) klass;
208 gstbasesrc_class = (GstBaseSrcClass *) klass;
209 gstaudiosrc_class = (GstAudioSrcClass *) klass;
211 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasrc_finalize);
212 gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasrc_get_property);
213 gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasrc_set_property);
215 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
217 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
218 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
219 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
220 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
221 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
222 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
223 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
225 g_object_class_install_property (gobject_class, PROP_DEVICE,
226 g_param_spec_string ("device", "Device",
227 "ALSA device, as defined in an asound configuration file",
228 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
230 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
231 g_param_spec_string ("device-name", "Device name",
232 "Human-readable name of the sound device",
233 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
237 gst_alsasrc_set_property (GObject * object, guint prop_id,
238 const GValue * value, GParamSpec * pspec)
242 src = GST_ALSA_SRC (object);
246 g_free (src->device);
247 src->device = g_value_dup_string (value);
248 if (src->device == NULL) {
249 src->device = g_strdup (DEFAULT_PROP_DEVICE);
253 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
259 gst_alsasrc_get_property (GObject * object, guint prop_id,
260 GValue * value, GParamSpec * pspec)
264 src = GST_ALSA_SRC (object);
268 g_value_set_string (value, src->device);
270 case PROP_DEVICE_NAME:
271 g_value_take_string (value,
272 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
273 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
276 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
282 gst_alsasrc_init (GstAlsaSrc * alsasrc, GstAlsaSrcClass * g_class)
284 GST_DEBUG_OBJECT (alsasrc, "initializing");
286 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
287 alsasrc->cached_caps = NULL;
289 alsasrc->alsa_lock = g_mutex_new ();
292 #define CHECK(call, error) \
294 if ((err = call) < 0) \
300 gst_alsasrc_getcaps (GstBaseSrc * bsrc)
302 GstElementClass *element_class;
303 GstPadTemplate *pad_template;
307 src = GST_ALSA_SRC (bsrc);
309 if (src->handle == NULL) {
310 GST_DEBUG_OBJECT (src, "device not open, using template caps");
311 return NULL; /* base class will get template caps for us */
314 if (src->cached_caps) {
315 GST_LOG_OBJECT (src, "Returning cached caps");
316 return gst_caps_ref (src->cached_caps);
319 element_class = GST_ELEMENT_GET_CLASS (src);
320 pad_template = gst_element_class_get_pad_template (element_class, "src");
321 g_return_val_if_fail (pad_template != NULL, NULL);
323 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
324 gst_pad_template_get_caps (pad_template));
327 src->cached_caps = gst_caps_ref (caps);
330 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
336 set_hwparams (GstAlsaSrc * alsa)
340 snd_pcm_hw_params_t *params;
342 snd_pcm_hw_params_malloc (¶ms);
344 /* choose all parameters */
345 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
346 /* set the interleaved read/write format */
347 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
349 /* set the sample format */
350 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
352 /* set the count of channels */
353 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
355 /* set the stream rate */
357 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
359 if (rrate != alsa->rate)
362 if (alsa->buffer_time != -1) {
363 /* set the buffer time */
364 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
365 &alsa->buffer_time, &dir), buffer_time);
367 if (alsa->period_time != -1) {
368 /* set the period time */
369 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
370 &alsa->period_time, &dir), period_time);
373 /* write the parameters to device */
374 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
376 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
379 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, &dir),
382 snd_pcm_hw_params_free (params);
388 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
389 ("Broken configuration for recording: no configurations available: %s",
390 snd_strerror (err)));
391 snd_pcm_hw_params_free (params);
396 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
397 ("Access type not available for recording: %s", snd_strerror (err)));
398 snd_pcm_hw_params_free (params);
403 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
404 ("Sample format not available for recording: %s", snd_strerror (err)));
405 snd_pcm_hw_params_free (params);
412 if ((alsa->channels) == 1)
413 msg = g_strdup (_("Could not open device for recording in mono mode."));
414 if ((alsa->channels) == 2)
415 msg = g_strdup (_("Could not open device for recording in stereo mode."));
416 if ((alsa->channels) > 2)
419 ("Could not open device for recording in %d-channel mode"),
421 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (msg), (snd_strerror (err)));
423 snd_pcm_hw_params_free (params);
428 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
429 ("Rate %iHz not available for recording: %s",
430 alsa->rate, snd_strerror (err)));
431 snd_pcm_hw_params_free (params);
436 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
437 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
438 snd_pcm_hw_params_free (params);
443 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
444 ("Unable to set buffer time %i for recording: %s",
445 alsa->buffer_time, snd_strerror (err)));
446 snd_pcm_hw_params_free (params);
451 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
452 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
453 snd_pcm_hw_params_free (params);
458 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
459 ("Unable to set period time %i for recording: %s", alsa->period_time,
460 snd_strerror (err)));
461 snd_pcm_hw_params_free (params);
466 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
467 ("Unable to get period size for recording: %s", snd_strerror (err)));
468 snd_pcm_hw_params_free (params);
473 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
474 ("Unable to set hw params for recording: %s", snd_strerror (err)));
475 snd_pcm_hw_params_free (params);
481 set_swparams (GstAlsaSrc * alsa)
484 snd_pcm_sw_params_t *params;
486 snd_pcm_sw_params_malloc (¶ms);
488 /* get the current swparams */
489 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
490 /* allow the transfer when at least period_size samples can be processed */
491 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
492 alsa->period_size), set_avail);
493 /* start the transfer on first read */
494 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
495 0), start_threshold);
497 #if GST_CHECK_ALSA_VERSION(1,0,16)
498 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
500 /* align all transfers to 1 sample */
501 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
504 /* write the parameters to the recording device */
505 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
507 snd_pcm_sw_params_free (params);
513 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
514 ("Unable to determine current swparams for playback: %s",
515 snd_strerror (err)));
516 snd_pcm_sw_params_free (params);
521 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
522 ("Unable to set start threshold mode for playback: %s",
523 snd_strerror (err)));
524 snd_pcm_sw_params_free (params);
529 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
530 ("Unable to set avail min for playback: %s", snd_strerror (err)));
531 snd_pcm_sw_params_free (params);
534 #if !GST_CHECK_ALSA_VERSION(1,0,16)
537 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
538 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
539 snd_pcm_sw_params_free (params);
545 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
546 ("Unable to set sw params for playback: %s", snd_strerror (err)));
547 snd_pcm_sw_params_free (params);
553 alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec)
555 switch (spec->type) {
556 case GST_BUFTYPE_LINEAR:
557 alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
558 spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
560 case GST_BUFTYPE_FLOAT:
561 switch (spec->format) {
563 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
566 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
569 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
572 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
578 case GST_BUFTYPE_A_LAW:
579 alsa->format = SND_PCM_FORMAT_A_LAW;
581 case GST_BUFTYPE_MU_LAW:
582 alsa->format = SND_PCM_FORMAT_MU_LAW;
588 alsa->rate = spec->rate;
589 alsa->channels = spec->channels;
590 alsa->buffer_time = spec->buffer_time;
591 alsa->period_time = spec->latency_time;
592 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
604 gst_alsasrc_open (GstAudioSrc * asrc)
609 alsa = GST_ALSA_SRC (asrc);
611 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
612 SND_PCM_NONBLOCK), open_error);
615 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
623 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
624 (_("Could not open audio device for recording. "
625 "Device is being used by another application.")),
626 ("Device '%s' is busy", alsa->device));
628 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
629 (_("Could not open audio device for recording.")),
630 ("Recording open error on device '%s': %s", alsa->device,
631 snd_strerror (err)));
638 gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
643 alsa = GST_ALSA_SRC (asrc);
645 if (!alsasrc_parse_spec (alsa, spec))
648 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
650 CHECK (set_hwparams (alsa), hw_params_failed);
651 CHECK (set_swparams (alsa), sw_params_failed);
652 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
654 alsa->bytes_per_sample = spec->bytes_per_sample;
655 spec->segsize = alsa->period_size * spec->bytes_per_sample;
656 spec->segtotal = alsa->buffer_size / alsa->period_size;
657 spec->silence_sample[0] = 0;
658 spec->silence_sample[1] = 0;
659 spec->silence_sample[2] = 0;
660 spec->silence_sample[3] = 0;
667 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
668 ("Error parsing spec"));
673 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
674 ("Could not set device to blocking: %s", snd_strerror (err)));
679 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
680 ("Setting of hwparams failed: %s", snd_strerror (err)));
685 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
686 ("Setting of swparams failed: %s", snd_strerror (err)));
691 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
692 ("Prepare failed: %s", snd_strerror (err)));
698 gst_alsasrc_unprepare (GstAudioSrc * asrc)
703 alsa = GST_ALSA_SRC (asrc);
705 CHECK (snd_pcm_drop (alsa->handle), drop);
707 CHECK (snd_pcm_hw_free (alsa->handle), hw_free);
709 CHECK (snd_pcm_nonblock (alsa->handle, 1), non_block);
716 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
717 ("Could not drop samples: %s", snd_strerror (err)));
722 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
723 ("Could not free hw params: %s", snd_strerror (err)));
728 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
729 ("Could not set device to nonblocking: %s", snd_strerror (err)));
735 gst_alsasrc_close (GstAudioSrc * asrc)
737 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
739 snd_pcm_close (alsa->handle);
743 gst_alsa_mixer_free (alsa->mixer);
747 gst_caps_replace (&alsa->cached_caps, NULL);
753 * Underrun and suspend recovery
756 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
758 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
760 if (err == -EPIPE) { /* under-run */
761 err = snd_pcm_prepare (handle);
763 GST_WARNING_OBJECT (alsa,
764 "Can't recovery from underrun, prepare failed: %s",
767 } else if (err == -ESTRPIPE) {
768 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
769 g_usleep (100); /* wait until the suspend flag is released */
772 err = snd_pcm_prepare (handle);
774 GST_WARNING_OBJECT (alsa,
775 "Can't recovery from suspend, prepare failed: %s",
784 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
791 alsa = GST_ALSA_SRC (asrc);
793 cptr = length / alsa->bytes_per_sample;
796 GST_ALSA_SRC_LOCK (asrc);
798 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
799 if (err == -EAGAIN) {
800 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
802 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
808 ptr += err * alsa->channels;
811 GST_ALSA_SRC_UNLOCK (asrc);
813 return length - cptr;
817 GST_ALSA_SRC_UNLOCK (asrc);
818 return length; /* skip one period */
823 gst_alsasrc_delay (GstAudioSrc * asrc)
826 snd_pcm_sframes_t delay;
829 alsa = GST_ALSA_SRC (asrc);
831 res = snd_pcm_delay (alsa->handle, &delay);
832 if (G_UNLIKELY (res < 0)) {
833 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
837 return CLAMP (delay, 0, alsa->buffer_size);
841 gst_alsasrc_reset (GstAudioSrc * asrc)
846 alsa = GST_ALSA_SRC (asrc);
848 GST_ALSA_SRC_LOCK (asrc);
849 GST_DEBUG_OBJECT (alsa, "drop");
850 CHECK (snd_pcm_drop (alsa->handle), drop_error);
851 GST_DEBUG_OBJECT (alsa, "prepare");
852 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
853 GST_DEBUG_OBJECT (alsa, "reset done");
854 GST_ALSA_SRC_UNLOCK (asrc);
861 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
863 GST_ALSA_SRC_UNLOCK (asrc);
868 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
870 GST_ALSA_SRC_UNLOCK (asrc);