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, GstCaps * filter);
82 static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
83 static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
84 GstAudioRingBufferSpec * 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, "
108 "format = (string) " GST_AUDIO_FORMATS_ALL ", "
109 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
113 gst_alsasrc_finalize (GObject * object)
115 GstAlsaSrc *src = GST_ALSA_SRC (object);
117 g_free (src->device);
118 g_mutex_free (src->alsa_lock);
120 G_OBJECT_CLASS (parent_class)->finalize (object);
124 gst_alsasrc_init_interfaces (GType type)
126 static const GInterfaceInfo mixer_iface_info = {
127 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
132 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
134 gst_alsa_type_add_device_property_probe_interface (type);
138 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
140 GObjectClass *gobject_class;
141 GstElementClass *gstelement_class;
142 GstBaseSrcClass *gstbasesrc_class;
143 GstAudioSrcClass *gstaudiosrc_class;
145 gobject_class = (GObjectClass *) klass;
146 gstelement_class = (GstElementClass *) klass;
147 gstbasesrc_class = (GstBaseSrcClass *) klass;
148 gstaudiosrc_class = (GstAudioSrcClass *) klass;
150 gobject_class->finalize = gst_alsasrc_finalize;
151 gobject_class->get_property = gst_alsasrc_get_property;
152 gobject_class->set_property = gst_alsasrc_set_property;
154 gst_element_class_set_details_simple (gstelement_class,
155 "Audio source (ALSA)", "Source/Audio",
156 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
158 gst_element_class_add_pad_template (gstelement_class,
159 gst_static_pad_template_get (&alsasrc_src_factory));
161 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
163 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
164 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
165 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
166 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
167 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
168 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
169 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
171 g_object_class_install_property (gobject_class, PROP_DEVICE,
172 g_param_spec_string ("device", "Device",
173 "ALSA device, as defined in an asound configuration file",
174 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
176 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
177 g_param_spec_string ("device-name", "Device name",
178 "Human-readable name of the sound device",
179 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
181 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
182 g_param_spec_string ("card-name", "Card name",
183 "Human-readable name of the sound card",
184 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
188 gst_alsasrc_set_property (GObject * object, guint prop_id,
189 const GValue * value, GParamSpec * pspec)
193 src = GST_ALSA_SRC (object);
197 g_free (src->device);
198 src->device = g_value_dup_string (value);
199 if (src->device == NULL) {
200 src->device = g_strdup (DEFAULT_PROP_DEVICE);
204 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
210 gst_alsasrc_get_property (GObject * object, guint prop_id,
211 GValue * value, GParamSpec * pspec)
215 src = GST_ALSA_SRC (object);
219 g_value_set_string (value, src->device);
221 case PROP_DEVICE_NAME:
222 g_value_take_string (value,
223 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
224 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
227 g_value_take_string (value,
228 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
229 src->device, SND_PCM_STREAM_CAPTURE));
232 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
238 gst_alsasrc_init (GstAlsaSrc * alsasrc)
240 GST_DEBUG_OBJECT (alsasrc, "initializing");
242 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
243 alsasrc->cached_caps = NULL;
245 alsasrc->alsa_lock = g_mutex_new ();
248 #define CHECK(call, error) \
250 if ((err = call) < 0) \
256 gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
258 GstElementClass *element_class;
259 GstPadTemplate *pad_template;
261 GstCaps *caps, *templ_caps;
263 src = GST_ALSA_SRC (bsrc);
265 if (src->handle == NULL) {
266 GST_DEBUG_OBJECT (src, "device not open, using template caps");
267 return GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
270 if (src->cached_caps) {
271 GST_LOG_OBJECT (src, "Returning cached caps");
273 return gst_caps_intersect_full (filter, src->cached_caps,
274 GST_CAPS_INTERSECT_FIRST);
276 return gst_caps_ref (src->cached_caps);
279 element_class = GST_ELEMENT_GET_CLASS (src);
280 pad_template = gst_element_class_get_pad_template (element_class, "src");
281 g_return_val_if_fail (pad_template != NULL, NULL);
283 templ_caps = gst_pad_template_get_caps (pad_template);
284 GST_INFO_OBJECT (src, "template caps %" GST_PTR_FORMAT, templ_caps);
286 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
288 gst_caps_unref (templ_caps);
291 src->cached_caps = gst_caps_ref (caps);
294 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
297 GstCaps *intersection;
300 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
301 gst_caps_unref (caps);
309 set_hwparams (GstAlsaSrc * alsa)
313 snd_pcm_hw_params_t *params;
315 snd_pcm_hw_params_malloc (¶ms);
317 /* choose all parameters */
318 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
319 /* set the interleaved read/write format */
320 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
322 /* set the sample format */
323 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
325 /* set the count of channels */
326 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
328 /* set the stream rate */
330 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
332 if (rrate != alsa->rate)
335 if (alsa->buffer_time != -1) {
336 /* set the buffer time */
337 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
338 &alsa->buffer_time, NULL), buffer_time);
340 if (alsa->period_time != -1) {
341 /* set the period time */
342 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
343 &alsa->period_time, NULL), period_time);
346 /* write the parameters to device */
347 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
349 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
352 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
355 snd_pcm_hw_params_free (params);
361 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
362 ("Broken configuration for recording: no configurations available: %s",
363 snd_strerror (err)));
364 snd_pcm_hw_params_free (params);
369 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
370 ("Access type not available for recording: %s", snd_strerror (err)));
371 snd_pcm_hw_params_free (params);
376 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
377 ("Sample format not available for recording: %s", snd_strerror (err)));
378 snd_pcm_hw_params_free (params);
385 if ((alsa->channels) == 1)
386 msg = g_strdup (_("Could not open device for recording in mono mode."));
387 if ((alsa->channels) == 2)
388 msg = g_strdup (_("Could not open device for recording in stereo mode."));
389 if ((alsa->channels) > 2)
392 ("Could not open device for recording in %d-channel mode"),
394 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
395 ("%s", snd_strerror (err)));
397 snd_pcm_hw_params_free (params);
402 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
403 ("Rate %iHz not available for recording: %s",
404 alsa->rate, snd_strerror (err)));
405 snd_pcm_hw_params_free (params);
410 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
411 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
412 snd_pcm_hw_params_free (params);
417 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
418 ("Unable to set buffer time %i for recording: %s",
419 alsa->buffer_time, snd_strerror (err)));
420 snd_pcm_hw_params_free (params);
425 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
426 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
427 snd_pcm_hw_params_free (params);
432 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
433 ("Unable to set period time %i for recording: %s", alsa->period_time,
434 snd_strerror (err)));
435 snd_pcm_hw_params_free (params);
440 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
441 ("Unable to get period size for recording: %s", snd_strerror (err)));
442 snd_pcm_hw_params_free (params);
447 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
448 ("Unable to set hw params for recording: %s", snd_strerror (err)));
449 snd_pcm_hw_params_free (params);
455 set_swparams (GstAlsaSrc * alsa)
458 snd_pcm_sw_params_t *params;
460 snd_pcm_sw_params_malloc (¶ms);
462 /* get the current swparams */
463 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
464 /* allow the transfer when at least period_size samples can be processed */
465 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
466 alsa->period_size), set_avail);
467 /* start the transfer on first read */
468 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
469 0), start_threshold);
471 #if GST_CHECK_ALSA_VERSION(1,0,16)
472 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
474 /* align all transfers to 1 sample */
475 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
478 /* write the parameters to the recording device */
479 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
481 snd_pcm_sw_params_free (params);
487 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
488 ("Unable to determine current swparams for playback: %s",
489 snd_strerror (err)));
490 snd_pcm_sw_params_free (params);
495 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
496 ("Unable to set start threshold mode for playback: %s",
497 snd_strerror (err)));
498 snd_pcm_sw_params_free (params);
503 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
504 ("Unable to set avail min for playback: %s", snd_strerror (err)));
505 snd_pcm_sw_params_free (params);
508 #if !GST_CHECK_ALSA_VERSION(1,0,16)
511 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
512 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
513 snd_pcm_sw_params_free (params);
519 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
520 ("Unable to set sw params for playback: %s", snd_strerror (err)));
521 snd_pcm_sw_params_free (params);
527 alsasrc_parse_spec (GstAlsaSrc * alsa, GstAudioRingBufferSpec * spec)
529 switch (spec->type) {
530 case GST_BUFTYPE_RAW:
531 switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
532 case GST_AUDIO_FORMAT_U8:
533 alsa->format = SND_PCM_FORMAT_U8;
535 case GST_AUDIO_FORMAT_S8:
536 alsa->format = SND_PCM_FORMAT_S8;
538 case GST_AUDIO_FORMAT_S16LE:
539 alsa->format = SND_PCM_FORMAT_S16_LE;
541 case GST_AUDIO_FORMAT_S16BE:
542 alsa->format = SND_PCM_FORMAT_S16_BE;
544 case GST_AUDIO_FORMAT_U16LE:
545 alsa->format = SND_PCM_FORMAT_U16_LE;
547 case GST_AUDIO_FORMAT_U16BE:
548 alsa->format = SND_PCM_FORMAT_U16_BE;
550 case GST_AUDIO_FORMAT_S24_32LE:
551 alsa->format = SND_PCM_FORMAT_S24_LE;
553 case GST_AUDIO_FORMAT_S24_32BE:
554 alsa->format = SND_PCM_FORMAT_S24_BE;
556 case GST_AUDIO_FORMAT_U24_32LE:
557 alsa->format = SND_PCM_FORMAT_U24_LE;
559 case GST_AUDIO_FORMAT_U24_32BE:
560 alsa->format = SND_PCM_FORMAT_U24_BE;
562 case GST_AUDIO_FORMAT_S32LE:
563 alsa->format = SND_PCM_FORMAT_S32_LE;
565 case GST_AUDIO_FORMAT_S32BE:
566 alsa->format = SND_PCM_FORMAT_S32_BE;
568 case GST_AUDIO_FORMAT_U32LE:
569 alsa->format = SND_PCM_FORMAT_U32_LE;
571 case GST_AUDIO_FORMAT_U32BE:
572 alsa->format = SND_PCM_FORMAT_U32_BE;
574 case GST_AUDIO_FORMAT_S24LE:
575 alsa->format = SND_PCM_FORMAT_S24_3LE;
577 case GST_AUDIO_FORMAT_S24BE:
578 alsa->format = SND_PCM_FORMAT_S24_3BE;
580 case GST_AUDIO_FORMAT_U24LE:
581 alsa->format = SND_PCM_FORMAT_U24_3LE;
583 case GST_AUDIO_FORMAT_U24BE:
584 alsa->format = SND_PCM_FORMAT_U24_3BE;
586 case GST_AUDIO_FORMAT_S20LE:
587 alsa->format = SND_PCM_FORMAT_S20_3LE;
589 case GST_AUDIO_FORMAT_S20BE:
590 alsa->format = SND_PCM_FORMAT_S20_3BE;
592 case GST_AUDIO_FORMAT_U20LE:
593 alsa->format = SND_PCM_FORMAT_U20_3LE;
595 case GST_AUDIO_FORMAT_U20BE:
596 alsa->format = SND_PCM_FORMAT_U20_3BE;
598 case GST_AUDIO_FORMAT_S18LE:
599 alsa->format = SND_PCM_FORMAT_S18_3LE;
601 case GST_AUDIO_FORMAT_S18BE:
602 alsa->format = SND_PCM_FORMAT_S18_3BE;
604 case GST_AUDIO_FORMAT_U18LE:
605 alsa->format = SND_PCM_FORMAT_U18_3LE;
607 case GST_AUDIO_FORMAT_U18BE:
608 alsa->format = SND_PCM_FORMAT_U18_3BE;
610 case GST_AUDIO_FORMAT_F32LE:
611 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
613 case GST_AUDIO_FORMAT_F32BE:
614 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
616 case GST_AUDIO_FORMAT_F64LE:
617 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
619 case GST_AUDIO_FORMAT_F64BE:
620 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
626 case GST_BUFTYPE_A_LAW:
627 alsa->format = SND_PCM_FORMAT_A_LAW;
629 case GST_BUFTYPE_MU_LAW:
630 alsa->format = SND_PCM_FORMAT_MU_LAW;
636 alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
637 alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
638 alsa->buffer_time = spec->buffer_time;
639 alsa->period_time = spec->latency_time;
640 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
652 gst_alsasrc_open (GstAudioSrc * asrc)
657 alsa = GST_ALSA_SRC (asrc);
659 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
660 SND_PCM_NONBLOCK), open_error);
663 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
671 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
672 (_("Could not open audio device for recording. "
673 "Device is being used by another application.")),
674 ("Device '%s' is busy", alsa->device));
676 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
677 (_("Could not open audio device for recording.")),
678 ("Recording open error on device '%s': %s", alsa->device,
679 snd_strerror (err)));
686 gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
691 alsa = GST_ALSA_SRC (asrc);
693 if (!alsasrc_parse_spec (alsa, spec))
696 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
698 CHECK (set_hwparams (alsa), hw_params_failed);
699 CHECK (set_swparams (alsa), sw_params_failed);
700 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
702 alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
703 spec->segsize = alsa->period_size * alsa->bpf;
704 spec->segtotal = alsa->buffer_size / alsa->period_size;
711 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
712 ("Error parsing spec"));
717 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
718 ("Could not set device to blocking: %s", snd_strerror (err)));
723 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
724 ("Setting of hwparams failed: %s", snd_strerror (err)));
729 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
730 ("Setting of swparams failed: %s", snd_strerror (err)));
735 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
736 ("Prepare failed: %s", snd_strerror (err)));
742 gst_alsasrc_unprepare (GstAudioSrc * asrc)
746 alsa = GST_ALSA_SRC (asrc);
748 snd_pcm_drop (alsa->handle);
749 snd_pcm_hw_free (alsa->handle);
750 snd_pcm_nonblock (alsa->handle, 1);
756 gst_alsasrc_close (GstAudioSrc * asrc)
758 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
760 snd_pcm_close (alsa->handle);
764 gst_alsa_mixer_free (alsa->mixer);
768 gst_caps_replace (&alsa->cached_caps, NULL);
774 * Underrun and suspend recovery
777 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
779 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
781 if (err == -EPIPE) { /* under-run */
782 err = snd_pcm_prepare (handle);
784 GST_WARNING_OBJECT (alsa,
785 "Can't recovery from underrun, prepare failed: %s",
788 } else if (err == -ESTRPIPE) {
789 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
790 g_usleep (100); /* wait until the suspend flag is released */
793 err = snd_pcm_prepare (handle);
795 GST_WARNING_OBJECT (alsa,
796 "Can't recovery from suspend, prepare failed: %s",
805 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
812 alsa = GST_ALSA_SRC (asrc);
814 cptr = length / alsa->bpf;
817 GST_ALSA_SRC_LOCK (asrc);
819 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
820 if (err == -EAGAIN) {
821 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
823 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
829 ptr += err * alsa->channels;
832 GST_ALSA_SRC_UNLOCK (asrc);
834 return length - (cptr * alsa->bpf);
838 GST_ALSA_SRC_UNLOCK (asrc);
839 return length; /* skip one period */
844 gst_alsasrc_delay (GstAudioSrc * asrc)
847 snd_pcm_sframes_t delay;
850 alsa = GST_ALSA_SRC (asrc);
852 res = snd_pcm_delay (alsa->handle, &delay);
853 if (G_UNLIKELY (res < 0)) {
854 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
858 return CLAMP (delay, 0, alsa->buffer_size);
862 gst_alsasrc_reset (GstAudioSrc * asrc)
867 alsa = GST_ALSA_SRC (asrc);
869 GST_ALSA_SRC_LOCK (asrc);
870 GST_DEBUG_OBJECT (alsa, "drop");
871 CHECK (snd_pcm_drop (alsa->handle), drop_error);
872 GST_DEBUG_OBJECT (alsa, "prepare");
873 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
874 GST_DEBUG_OBJECT (alsa, "reset done");
875 GST_ALSA_SRC_UNLOCK (asrc);
882 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
884 GST_ALSA_SRC_UNLOCK (asrc);
889 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
891 GST_ALSA_SRC_UNLOCK (asrc);