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-lpcm, "
109 "endianness = (int) { 1234, 4321 }, "
110 "signed = (boolean) { TRUE, FALSE }, "
113 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ];"
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 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
134 "signed = (boolean) { TRUE, FALSE }, "
137 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
139 "signed = (boolean) { TRUE, FALSE }, "
142 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
146 gst_alsasrc_finalize (GObject * object)
148 GstAlsaSrc *src = GST_ALSA_SRC (object);
150 g_free (src->device);
151 g_mutex_free (src->alsa_lock);
153 G_OBJECT_CLASS (parent_class)->finalize (object);
157 gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type)
159 /* only support this one interface (wrapped by GstImplementsInterface) */
160 g_assert (interface_type == GST_TYPE_MIXER);
162 return gst_alsasrc_mixer_supported (this, interface_type);
166 gst_implements_interface_init (GstImplementsInterfaceClass * klass)
168 klass->supported = (gpointer) gst_alsasrc_interface_supported;
172 gst_alsasrc_init_interfaces (GType type)
174 static const GInterfaceInfo implements_iface_info = {
175 (GInterfaceInitFunc) gst_implements_interface_init,
179 static const GInterfaceInfo mixer_iface_info = {
180 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
185 g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
186 &implements_iface_info);
187 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
189 gst_alsa_type_add_device_property_probe_interface (type);
193 gst_alsasrc_base_init (gpointer g_class)
195 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
197 gst_element_class_set_details_simple (element_class,
198 "Audio source (ALSA)", "Source/Audio",
199 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
201 gst_element_class_add_static_pad_template (element_class,
202 &alsasrc_src_factory);
206 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
208 GObjectClass *gobject_class;
209 GstBaseSrcClass *gstbasesrc_class;
210 GstAudioSrcClass *gstaudiosrc_class;
212 gobject_class = (GObjectClass *) klass;
213 gstbasesrc_class = (GstBaseSrcClass *) klass;
214 gstaudiosrc_class = (GstAudioSrcClass *) klass;
216 gobject_class->finalize = gst_alsasrc_finalize;
217 gobject_class->get_property = gst_alsasrc_get_property;
218 gobject_class->set_property = gst_alsasrc_set_property;
220 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
222 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
223 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
224 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
225 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
226 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
227 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
228 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
230 g_object_class_install_property (gobject_class, PROP_DEVICE,
231 g_param_spec_string ("device", "Device",
232 "ALSA device, as defined in an asound configuration file",
233 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
235 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
236 g_param_spec_string ("device-name", "Device name",
237 "Human-readable name of the sound device",
238 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
240 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
241 g_param_spec_string ("card-name", "Card name",
242 "Human-readable name of the sound card",
243 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
247 gst_alsasrc_set_property (GObject * object, guint prop_id,
248 const GValue * value, GParamSpec * pspec)
252 src = GST_ALSA_SRC (object);
256 g_free (src->device);
257 src->device = g_value_dup_string (value);
258 if (src->device == NULL) {
259 src->device = g_strdup (DEFAULT_PROP_DEVICE);
263 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
269 gst_alsasrc_get_property (GObject * object, guint prop_id,
270 GValue * value, GParamSpec * pspec)
274 src = GST_ALSA_SRC (object);
278 g_value_set_string (value, src->device);
280 case PROP_DEVICE_NAME:
281 g_value_take_string (value,
282 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
283 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
286 g_value_take_string (value,
287 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
288 src->device, SND_PCM_STREAM_CAPTURE));
291 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
297 gst_alsasrc_init (GstAlsaSrc * alsasrc, GstAlsaSrcClass * g_class)
299 GST_DEBUG_OBJECT (alsasrc, "initializing");
301 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
302 alsasrc->cached_caps = NULL;
304 alsasrc->alsa_lock = g_mutex_new ();
307 #define CHECK(call, error) \
309 if ((err = call) < 0) \
315 gst_alsasrc_getcaps (GstBaseSrc * bsrc)
317 GstElementClass *element_class;
318 GstPadTemplate *pad_template;
322 src = GST_ALSA_SRC (bsrc);
324 if (src->handle == NULL) {
325 GST_DEBUG_OBJECT (src, "device not open, using template caps");
326 return NULL; /* base class will get template caps for us */
329 if (src->cached_caps) {
330 GST_LOG_OBJECT (src, "Returning cached caps");
331 return gst_caps_ref (src->cached_caps);
334 element_class = GST_ELEMENT_GET_CLASS (src);
335 pad_template = gst_element_class_get_pad_template (element_class, "src");
336 g_return_val_if_fail (pad_template != NULL, NULL);
338 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
339 gst_pad_template_get_caps (pad_template));
342 src->cached_caps = gst_caps_ref (caps);
345 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
351 set_hwparams (GstAlsaSrc * alsa)
355 snd_pcm_hw_params_t *params;
357 snd_pcm_hw_params_malloc (¶ms);
359 /* choose all parameters */
360 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
361 /* set the interleaved read/write format */
362 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
364 /* set the sample format */
365 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
367 /* set the count of channels */
368 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
370 /* set the stream rate */
372 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
374 if (rrate != alsa->rate)
377 if (alsa->buffer_time != -1) {
378 /* set the buffer time */
379 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
380 &alsa->buffer_time, NULL), buffer_time);
382 if (alsa->period_time != -1) {
383 /* set the period time */
384 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
385 &alsa->period_time, NULL), period_time);
388 /* write the parameters to device */
389 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
391 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
394 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
397 snd_pcm_hw_params_free (params);
403 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
404 ("Broken configuration for recording: no configurations available: %s",
405 snd_strerror (err)));
406 snd_pcm_hw_params_free (params);
411 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
412 ("Access type not available for recording: %s", snd_strerror (err)));
413 snd_pcm_hw_params_free (params);
418 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
419 ("Sample format not available for recording: %s", snd_strerror (err)));
420 snd_pcm_hw_params_free (params);
427 if ((alsa->channels) == 1)
428 msg = g_strdup (_("Could not open device for recording in mono mode."));
429 if ((alsa->channels) == 2)
430 msg = g_strdup (_("Could not open device for recording in stereo mode."));
431 if ((alsa->channels) > 2)
434 ("Could not open device for recording in %d-channel mode"),
436 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
437 ("%s", snd_strerror (err)));
439 snd_pcm_hw_params_free (params);
444 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
445 ("Rate %iHz not available for recording: %s",
446 alsa->rate, snd_strerror (err)));
447 snd_pcm_hw_params_free (params);
452 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
453 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
454 snd_pcm_hw_params_free (params);
459 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
460 ("Unable to set buffer time %i for recording: %s",
461 alsa->buffer_time, snd_strerror (err)));
462 snd_pcm_hw_params_free (params);
467 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
468 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
469 snd_pcm_hw_params_free (params);
474 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
475 ("Unable to set period time %i for recording: %s", alsa->period_time,
476 snd_strerror (err)));
477 snd_pcm_hw_params_free (params);
482 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
483 ("Unable to get period size for recording: %s", snd_strerror (err)));
484 snd_pcm_hw_params_free (params);
489 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
490 ("Unable to set hw params for recording: %s", snd_strerror (err)));
491 snd_pcm_hw_params_free (params);
497 set_swparams (GstAlsaSrc * alsa)
500 snd_pcm_sw_params_t *params;
502 snd_pcm_sw_params_malloc (¶ms);
504 /* get the current swparams */
505 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
506 /* allow the transfer when at least period_size samples can be processed */
507 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
508 alsa->period_size), set_avail);
509 /* start the transfer on first read */
510 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
511 0), start_threshold);
513 #if GST_CHECK_ALSA_VERSION(1,0,16)
514 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
516 /* align all transfers to 1 sample */
517 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
520 /* write the parameters to the recording device */
521 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
523 snd_pcm_sw_params_free (params);
529 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
530 ("Unable to determine current swparams for playback: %s",
531 snd_strerror (err)));
532 snd_pcm_sw_params_free (params);
537 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
538 ("Unable to set start threshold mode for playback: %s",
539 snd_strerror (err)));
540 snd_pcm_sw_params_free (params);
545 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
546 ("Unable to set avail min for playback: %s", snd_strerror (err)));
547 snd_pcm_sw_params_free (params);
550 #if !GST_CHECK_ALSA_VERSION(1,0,16)
553 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
554 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
555 snd_pcm_sw_params_free (params);
561 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
562 ("Unable to set sw params for playback: %s", snd_strerror (err)));
563 snd_pcm_sw_params_free (params);
569 alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec)
571 switch (spec->type) {
572 case GST_BUFTYPE_LINEAR:
573 alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
574 spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
576 case GST_BUFTYPE_FLOAT:
577 switch (spec->format) {
579 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
582 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
585 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
588 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
594 case GST_BUFTYPE_A_LAW:
595 alsa->format = SND_PCM_FORMAT_A_LAW;
597 case GST_BUFTYPE_MU_LAW:
598 alsa->format = SND_PCM_FORMAT_MU_LAW;
604 alsa->rate = spec->rate;
605 alsa->channels = spec->channels;
606 alsa->buffer_time = spec->buffer_time;
607 alsa->period_time = spec->latency_time;
608 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
620 gst_alsasrc_open (GstAudioSrc * asrc)
625 alsa = GST_ALSA_SRC (asrc);
627 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
628 SND_PCM_NONBLOCK), open_error);
631 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
639 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
640 (_("Could not open audio device for recording. "
641 "Device is being used by another application.")),
642 ("Device '%s' is busy", alsa->device));
644 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
645 (_("Could not open audio device for recording.")),
646 ("Recording open error on device '%s': %s", alsa->device,
647 snd_strerror (err)));
654 gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
659 alsa = GST_ALSA_SRC (asrc);
661 if (!alsasrc_parse_spec (alsa, spec))
664 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
666 CHECK (set_hwparams (alsa), hw_params_failed);
667 CHECK (set_swparams (alsa), sw_params_failed);
668 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
670 alsa->bytes_per_sample = spec->bytes_per_sample;
671 spec->segsize = alsa->period_size * spec->bytes_per_sample;
672 spec->segtotal = alsa->buffer_size / alsa->period_size;
673 spec->silence_sample[0] = 0;
674 spec->silence_sample[1] = 0;
675 spec->silence_sample[2] = 0;
676 spec->silence_sample[3] = 0;
683 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
684 ("Error parsing spec"));
689 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
690 ("Could not set device to blocking: %s", snd_strerror (err)));
695 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
696 ("Setting of hwparams failed: %s", snd_strerror (err)));
701 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
702 ("Setting of swparams failed: %s", snd_strerror (err)));
707 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
708 ("Prepare failed: %s", snd_strerror (err)));
714 gst_alsasrc_unprepare (GstAudioSrc * asrc)
718 alsa = GST_ALSA_SRC (asrc);
720 snd_pcm_drop (alsa->handle);
721 snd_pcm_hw_free (alsa->handle);
722 snd_pcm_nonblock (alsa->handle, 1);
728 gst_alsasrc_close (GstAudioSrc * asrc)
730 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
732 snd_pcm_close (alsa->handle);
736 gst_alsa_mixer_free (alsa->mixer);
740 gst_caps_replace (&alsa->cached_caps, NULL);
746 * Underrun and suspend recovery
749 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
751 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
753 if (err == -EPIPE) { /* under-run */
754 err = snd_pcm_prepare (handle);
756 GST_WARNING_OBJECT (alsa,
757 "Can't recovery from underrun, prepare failed: %s",
760 } else if (err == -ESTRPIPE) {
761 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
762 g_usleep (100); /* wait until the suspend flag is released */
765 err = snd_pcm_prepare (handle);
767 GST_WARNING_OBJECT (alsa,
768 "Can't recovery from suspend, prepare failed: %s",
777 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
784 alsa = GST_ALSA_SRC (asrc);
786 cptr = length / alsa->bytes_per_sample;
789 GST_ALSA_SRC_LOCK (asrc);
791 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
792 if (err == -EAGAIN) {
793 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
795 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
801 ptr += err * alsa->channels;
804 GST_ALSA_SRC_UNLOCK (asrc);
806 return length - (cptr * alsa->bytes_per_sample);
810 GST_ALSA_SRC_UNLOCK (asrc);
811 return length; /* skip one period */
816 gst_alsasrc_delay (GstAudioSrc * asrc)
819 snd_pcm_sframes_t delay;
822 alsa = GST_ALSA_SRC (asrc);
824 res = snd_pcm_delay (alsa->handle, &delay);
825 if (G_UNLIKELY (res < 0)) {
826 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
830 return CLAMP (delay, 0, alsa->buffer_size);
834 gst_alsasrc_reset (GstAudioSrc * asrc)
839 alsa = GST_ALSA_SRC (asrc);
841 GST_ALSA_SRC_LOCK (asrc);
842 GST_DEBUG_OBJECT (alsa, "drop");
843 CHECK (snd_pcm_drop (alsa->handle), drop_error);
844 GST_DEBUG_OBJECT (alsa, "prepare");
845 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
846 GST_DEBUG_OBJECT (alsa, "reset done");
847 GST_ALSA_SRC_UNLOCK (asrc);
854 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
856 GST_ALSA_SRC_UNLOCK (asrc);
861 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
863 GST_ALSA_SRC_UNLOCK (asrc);