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);
69 #define gst_alsasrc_parent_class parent_class
70 G_DEFINE_TYPE_WITH_CODE (GstAlsaSrc, gst_alsasrc,
71 GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces (g_define_type_id));
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, GstCaps * filter);
83 static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
84 static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
85 GstAudioRingBufferSpec * 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, "
109 "format = (string) " GST_AUDIO_FORMATS_ALL ", "
110 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
114 gst_alsasrc_finalize (GObject * object)
116 GstAlsaSrc *src = GST_ALSA_SRC (object);
118 g_free (src->device);
119 g_mutex_free (src->alsa_lock);
121 G_OBJECT_CLASS (parent_class)->finalize (object);
125 gst_alsasrc_init_interfaces (GType type)
127 static const GInterfaceInfo mixer_iface_info = {
128 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
133 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
135 gst_alsa_type_add_device_property_probe_interface (type);
139 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
141 GObjectClass *gobject_class;
142 GstElementClass *gstelement_class;
143 GstBaseSrcClass *gstbasesrc_class;
144 GstAudioSrcClass *gstaudiosrc_class;
146 gobject_class = (GObjectClass *) klass;
147 gstelement_class = (GstElementClass *) klass;
148 gstbasesrc_class = (GstBaseSrcClass *) klass;
149 gstaudiosrc_class = (GstAudioSrcClass *) klass;
151 gobject_class->finalize = gst_alsasrc_finalize;
152 gobject_class->get_property = gst_alsasrc_get_property;
153 gobject_class->set_property = gst_alsasrc_set_property;
155 gst_element_class_set_details_simple (gstelement_class,
156 "Audio source (ALSA)", "Source/Audio",
157 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
159 gst_element_class_add_pad_template (gstelement_class,
160 gst_static_pad_template_get (&alsasrc_src_factory));
162 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
164 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
165 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
166 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
167 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
168 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
169 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
170 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
172 g_object_class_install_property (gobject_class, PROP_DEVICE,
173 g_param_spec_string ("device", "Device",
174 "ALSA device, as defined in an asound configuration file",
175 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
177 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
178 g_param_spec_string ("device-name", "Device name",
179 "Human-readable name of the sound device",
180 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
182 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
183 g_param_spec_string ("card-name", "Card name",
184 "Human-readable name of the sound card",
185 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
189 gst_alsasrc_set_property (GObject * object, guint prop_id,
190 const GValue * value, GParamSpec * pspec)
194 src = GST_ALSA_SRC (object);
198 g_free (src->device);
199 src->device = g_value_dup_string (value);
200 if (src->device == NULL) {
201 src->device = g_strdup (DEFAULT_PROP_DEVICE);
205 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
211 gst_alsasrc_get_property (GObject * object, guint prop_id,
212 GValue * value, GParamSpec * pspec)
216 src = GST_ALSA_SRC (object);
220 g_value_set_string (value, src->device);
222 case PROP_DEVICE_NAME:
223 g_value_take_string (value,
224 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
225 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
228 g_value_take_string (value,
229 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
230 src->device, SND_PCM_STREAM_CAPTURE));
233 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
239 gst_alsasrc_init (GstAlsaSrc * alsasrc)
241 GST_DEBUG_OBJECT (alsasrc, "initializing");
243 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
244 alsasrc->cached_caps = NULL;
246 alsasrc->alsa_lock = g_mutex_new ();
249 #define CHECK(call, error) \
251 if ((err = call) < 0) \
257 gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
259 GstElementClass *element_class;
260 GstPadTemplate *pad_template;
262 GstCaps *caps, *templ_caps;
264 src = GST_ALSA_SRC (bsrc);
266 if (src->handle == NULL) {
267 GST_DEBUG_OBJECT (src, "device not open, using template caps");
268 return GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
271 if (src->cached_caps) {
272 GST_LOG_OBJECT (src, "Returning cached caps");
274 return gst_caps_intersect_full (filter, src->cached_caps,
275 GST_CAPS_INTERSECT_FIRST);
277 return gst_caps_ref (src->cached_caps);
280 element_class = GST_ELEMENT_GET_CLASS (src);
281 pad_template = gst_element_class_get_pad_template (element_class, "src");
282 g_return_val_if_fail (pad_template != NULL, NULL);
284 templ_caps = gst_pad_template_get_caps (pad_template);
285 GST_INFO_OBJECT (src, "template caps %" GST_PTR_FORMAT, templ_caps);
287 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
289 gst_caps_unref (templ_caps);
292 src->cached_caps = gst_caps_ref (caps);
295 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
298 GstCaps *intersection;
301 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
302 gst_caps_unref (caps);
310 set_hwparams (GstAlsaSrc * alsa)
314 snd_pcm_hw_params_t *params;
316 snd_pcm_hw_params_malloc (¶ms);
318 /* choose all parameters */
319 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
320 /* set the interleaved read/write format */
321 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
323 /* set the sample format */
324 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
326 /* set the count of channels */
327 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
329 /* set the stream rate */
331 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
333 if (rrate != alsa->rate)
336 if (alsa->buffer_time != -1) {
337 /* set the buffer time */
338 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
339 &alsa->buffer_time, NULL), buffer_time);
341 if (alsa->period_time != -1) {
342 /* set the period time */
343 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
344 &alsa->period_time, NULL), period_time);
347 /* write the parameters to device */
348 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
350 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
353 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
356 snd_pcm_hw_params_free (params);
362 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
363 ("Broken configuration for recording: no configurations available: %s",
364 snd_strerror (err)));
365 snd_pcm_hw_params_free (params);
370 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
371 ("Access type not available for recording: %s", snd_strerror (err)));
372 snd_pcm_hw_params_free (params);
377 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
378 ("Sample format not available for recording: %s", snd_strerror (err)));
379 snd_pcm_hw_params_free (params);
386 if ((alsa->channels) == 1)
387 msg = g_strdup (_("Could not open device for recording in mono mode."));
388 if ((alsa->channels) == 2)
389 msg = g_strdup (_("Could not open device for recording in stereo mode."));
390 if ((alsa->channels) > 2)
393 ("Could not open device for recording in %d-channel mode"),
395 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
396 ("%s", snd_strerror (err)));
398 snd_pcm_hw_params_free (params);
403 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
404 ("Rate %iHz not available for recording: %s",
405 alsa->rate, snd_strerror (err)));
406 snd_pcm_hw_params_free (params);
411 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
412 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
413 snd_pcm_hw_params_free (params);
418 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
419 ("Unable to set buffer time %i for recording: %s",
420 alsa->buffer_time, snd_strerror (err)));
421 snd_pcm_hw_params_free (params);
426 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
427 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
428 snd_pcm_hw_params_free (params);
433 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
434 ("Unable to set period time %i for recording: %s", alsa->period_time,
435 snd_strerror (err)));
436 snd_pcm_hw_params_free (params);
441 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
442 ("Unable to get period size for recording: %s", snd_strerror (err)));
443 snd_pcm_hw_params_free (params);
448 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
449 ("Unable to set hw params for recording: %s", snd_strerror (err)));
450 snd_pcm_hw_params_free (params);
456 set_swparams (GstAlsaSrc * alsa)
459 snd_pcm_sw_params_t *params;
461 snd_pcm_sw_params_malloc (¶ms);
463 /* get the current swparams */
464 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
465 /* allow the transfer when at least period_size samples can be processed */
466 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
467 alsa->period_size), set_avail);
468 /* start the transfer on first read */
469 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
470 0), start_threshold);
472 #if GST_CHECK_ALSA_VERSION(1,0,16)
473 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
475 /* align all transfers to 1 sample */
476 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
479 /* write the parameters to the recording device */
480 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
482 snd_pcm_sw_params_free (params);
488 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
489 ("Unable to determine current swparams for playback: %s",
490 snd_strerror (err)));
491 snd_pcm_sw_params_free (params);
496 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
497 ("Unable to set start threshold mode for playback: %s",
498 snd_strerror (err)));
499 snd_pcm_sw_params_free (params);
504 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
505 ("Unable to set avail min for playback: %s", snd_strerror (err)));
506 snd_pcm_sw_params_free (params);
509 #if !GST_CHECK_ALSA_VERSION(1,0,16)
512 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
513 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
514 snd_pcm_sw_params_free (params);
520 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
521 ("Unable to set sw params for playback: %s", snd_strerror (err)));
522 snd_pcm_sw_params_free (params);
528 alsasrc_parse_spec (GstAlsaSrc * alsa, GstAudioRingBufferSpec * spec)
530 switch (spec->type) {
531 case GST_BUFTYPE_RAW:
532 switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
533 case GST_AUDIO_FORMAT_U8:
534 alsa->format = SND_PCM_FORMAT_U8;
536 case GST_AUDIO_FORMAT_S8:
537 alsa->format = SND_PCM_FORMAT_S8;
539 case GST_AUDIO_FORMAT_S16LE:
540 alsa->format = SND_PCM_FORMAT_S16_LE;
542 case GST_AUDIO_FORMAT_S16BE:
543 alsa->format = SND_PCM_FORMAT_S16_BE;
545 case GST_AUDIO_FORMAT_U16LE:
546 alsa->format = SND_PCM_FORMAT_U16_LE;
548 case GST_AUDIO_FORMAT_U16BE:
549 alsa->format = SND_PCM_FORMAT_U16_BE;
551 case GST_AUDIO_FORMAT_S24_32LE:
552 alsa->format = SND_PCM_FORMAT_S24_LE;
554 case GST_AUDIO_FORMAT_S24_32BE:
555 alsa->format = SND_PCM_FORMAT_S24_BE;
557 case GST_AUDIO_FORMAT_U24_32LE:
558 alsa->format = SND_PCM_FORMAT_U24_LE;
560 case GST_AUDIO_FORMAT_U24_32BE:
561 alsa->format = SND_PCM_FORMAT_U24_BE;
563 case GST_AUDIO_FORMAT_S32LE:
564 alsa->format = SND_PCM_FORMAT_S32_LE;
566 case GST_AUDIO_FORMAT_S32BE:
567 alsa->format = SND_PCM_FORMAT_S32_BE;
569 case GST_AUDIO_FORMAT_U32LE:
570 alsa->format = SND_PCM_FORMAT_U32_LE;
572 case GST_AUDIO_FORMAT_U32BE:
573 alsa->format = SND_PCM_FORMAT_U32_BE;
575 case GST_AUDIO_FORMAT_S24LE:
576 alsa->format = SND_PCM_FORMAT_S24_3LE;
578 case GST_AUDIO_FORMAT_S24BE:
579 alsa->format = SND_PCM_FORMAT_S24_3BE;
581 case GST_AUDIO_FORMAT_U24LE:
582 alsa->format = SND_PCM_FORMAT_U24_3LE;
584 case GST_AUDIO_FORMAT_U24BE:
585 alsa->format = SND_PCM_FORMAT_U24_3BE;
587 case GST_AUDIO_FORMAT_S20LE:
588 alsa->format = SND_PCM_FORMAT_S20_3LE;
590 case GST_AUDIO_FORMAT_S20BE:
591 alsa->format = SND_PCM_FORMAT_S20_3BE;
593 case GST_AUDIO_FORMAT_U20LE:
594 alsa->format = SND_PCM_FORMAT_U20_3LE;
596 case GST_AUDIO_FORMAT_U20BE:
597 alsa->format = SND_PCM_FORMAT_U20_3BE;
599 case GST_AUDIO_FORMAT_S18LE:
600 alsa->format = SND_PCM_FORMAT_S18_3LE;
602 case GST_AUDIO_FORMAT_S18BE:
603 alsa->format = SND_PCM_FORMAT_S18_3BE;
605 case GST_AUDIO_FORMAT_U18LE:
606 alsa->format = SND_PCM_FORMAT_U18_3LE;
608 case GST_AUDIO_FORMAT_U18BE:
609 alsa->format = SND_PCM_FORMAT_U18_3BE;
611 case GST_AUDIO_FORMAT_F32LE:
612 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
614 case GST_AUDIO_FORMAT_F32BE:
615 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
617 case GST_AUDIO_FORMAT_F64LE:
618 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
620 case GST_AUDIO_FORMAT_F64BE:
621 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
627 case GST_BUFTYPE_A_LAW:
628 alsa->format = SND_PCM_FORMAT_A_LAW;
630 case GST_BUFTYPE_MU_LAW:
631 alsa->format = SND_PCM_FORMAT_MU_LAW;
637 alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
638 alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
639 alsa->buffer_time = spec->buffer_time;
640 alsa->period_time = spec->latency_time;
641 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
653 gst_alsasrc_open (GstAudioSrc * asrc)
658 alsa = GST_ALSA_SRC (asrc);
660 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
661 SND_PCM_NONBLOCK), open_error);
664 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
672 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
673 (_("Could not open audio device for recording. "
674 "Device is being used by another application.")),
675 ("Device '%s' is busy", alsa->device));
677 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
678 (_("Could not open audio device for recording.")),
679 ("Recording open error on device '%s': %s", alsa->device,
680 snd_strerror (err)));
687 gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
692 alsa = GST_ALSA_SRC (asrc);
694 if (!alsasrc_parse_spec (alsa, spec))
697 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
699 CHECK (set_hwparams (alsa), hw_params_failed);
700 CHECK (set_swparams (alsa), sw_params_failed);
701 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
703 alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
704 spec->segsize = alsa->period_size * alsa->bpf;
705 spec->segtotal = alsa->buffer_size / alsa->period_size;
712 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
713 ("Error parsing spec"));
718 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
719 ("Could not set device to blocking: %s", snd_strerror (err)));
724 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
725 ("Setting of hwparams failed: %s", snd_strerror (err)));
730 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
731 ("Setting of swparams failed: %s", snd_strerror (err)));
736 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
737 ("Prepare failed: %s", snd_strerror (err)));
743 gst_alsasrc_unprepare (GstAudioSrc * asrc)
747 alsa = GST_ALSA_SRC (asrc);
749 snd_pcm_drop (alsa->handle);
750 snd_pcm_hw_free (alsa->handle);
751 snd_pcm_nonblock (alsa->handle, 1);
757 gst_alsasrc_close (GstAudioSrc * asrc)
759 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
761 snd_pcm_close (alsa->handle);
765 gst_alsa_mixer_free (alsa->mixer);
769 gst_caps_replace (&alsa->cached_caps, NULL);
775 * Underrun and suspend recovery
778 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
780 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
782 if (err == -EPIPE) { /* under-run */
783 err = snd_pcm_prepare (handle);
785 GST_WARNING_OBJECT (alsa,
786 "Can't recovery from underrun, prepare failed: %s",
789 } else if (err == -ESTRPIPE) {
790 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
791 g_usleep (100); /* wait until the suspend flag is released */
794 err = snd_pcm_prepare (handle);
796 GST_WARNING_OBJECT (alsa,
797 "Can't recovery from suspend, prepare failed: %s",
806 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
813 alsa = GST_ALSA_SRC (asrc);
815 cptr = length / alsa->bpf;
818 GST_ALSA_SRC_LOCK (asrc);
820 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
821 if (err == -EAGAIN) {
822 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
824 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
830 ptr += err * alsa->channels;
833 GST_ALSA_SRC_UNLOCK (asrc);
835 return length - (cptr * alsa->bpf);
839 GST_ALSA_SRC_UNLOCK (asrc);
840 return length; /* skip one period */
845 gst_alsasrc_delay (GstAudioSrc * asrc)
848 snd_pcm_sframes_t delay;
851 alsa = GST_ALSA_SRC (asrc);
853 res = snd_pcm_delay (alsa->handle, &delay);
854 if (G_UNLIKELY (res < 0)) {
855 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
859 return CLAMP (delay, 0, alsa->buffer_size);
863 gst_alsasrc_reset (GstAudioSrc * asrc)
868 alsa = GST_ALSA_SRC (asrc);
870 GST_ALSA_SRC_LOCK (asrc);
871 GST_DEBUG_OBJECT (alsa, "drop");
872 CHECK (snd_pcm_drop (alsa->handle), drop_error);
873 GST_DEBUG_OBJECT (alsa, "prepare");
874 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
875 GST_DEBUG_OBJECT (alsa, "reset done");
876 GST_ALSA_SRC_UNLOCK (asrc);
883 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
885 GST_ALSA_SRC_UNLOCK (asrc);
890 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
892 GST_ALSA_SRC_UNLOCK (asrc);