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 ""
69 static void gst_alsasrc_init_interfaces (GType type);
71 GST_BOILERPLATE_FULL (GstAlsaSrc, gst_alsasrc, GstAudioSrc,
72 GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces);
74 GST_IMPLEMENT_ALSA_MIXER_METHODS (GstAlsaSrc, gst_alsasrc_mixer);
76 static void gst_alsasrc_finalize (GObject * object);
77 static void gst_alsasrc_set_property (GObject * object,
78 guint prop_id, const GValue * value, GParamSpec * pspec);
79 static void gst_alsasrc_get_property (GObject * object,
80 guint prop_id, GValue * value, GParamSpec * pspec);
82 static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc);
84 static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
85 static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
86 GstRingBufferSpec * spec);
87 static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);
88 static gboolean gst_alsasrc_close (GstAudioSrc * asrc);
89 static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length);
90 static guint gst_alsasrc_delay (GstAudioSrc * asrc);
91 static void gst_alsasrc_reset (GstAudioSrc * asrc);
92 static GstStateChangeReturn gst_alsasrc_change_state (GstElement * element, GstStateChange transition);
94 /* AlsaSrc signals and args */
100 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
101 # define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
103 # define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
106 static GstStaticPadTemplate alsasrc_src_factory =
107 GST_STATIC_PAD_TEMPLATE ("src",
110 GST_STATIC_CAPS ("audio/x-lpcm, "
111 "endianness = (int) { 1234, 4321 }, "
112 "signed = (boolean) { TRUE, FALSE }, "
115 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ];"
117 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
118 "signed = (boolean) { TRUE, FALSE }, "
121 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
123 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
124 "signed = (boolean) { TRUE, FALSE }, "
127 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
129 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
130 "signed = (boolean) { TRUE, FALSE }, "
133 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
135 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
136 "signed = (boolean) { TRUE, FALSE }, "
139 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
141 "signed = (boolean) { TRUE, FALSE }, "
144 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
148 gst_alsasrc_finalize (GObject * object)
150 GstAlsaSrc *src = GST_ALSA_SRC (object);
152 g_free (src->device);
153 g_mutex_free (src->alsa_lock);
155 G_OBJECT_CLASS (parent_class)->finalize (object);
159 gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type)
161 /* only support this one interface (wrapped by GstImplementsInterface) */
162 g_assert (interface_type == GST_TYPE_MIXER);
164 return gst_alsasrc_mixer_supported (this, interface_type);
168 gst_implements_interface_init (GstImplementsInterfaceClass * klass)
170 klass->supported = (gpointer) gst_alsasrc_interface_supported;
174 gst_alsasrc_init_interfaces (GType type)
176 static const GInterfaceInfo implements_iface_info = {
177 (GInterfaceInitFunc) gst_implements_interface_init,
181 static const GInterfaceInfo mixer_iface_info = {
182 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
187 g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
188 &implements_iface_info);
189 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
191 gst_alsa_type_add_device_property_probe_interface (type);
195 gst_alsasrc_base_init (gpointer g_class)
197 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
199 gst_element_class_set_details_simple (element_class,
200 "Audio source (ALSA)", "Source/Audio",
201 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
203 gst_element_class_add_static_pad_template (element_class,
204 &alsasrc_src_factory);
208 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
210 GObjectClass *gobject_class;
211 GstBaseSrcClass *gstbasesrc_class;
212 GstAudioSrcClass *gstaudiosrc_class;
213 GstElementClass *gstelement_class;
215 gobject_class = (GObjectClass *) klass;
216 gstelement_class = (GstElementClass *) klass;
217 gstbasesrc_class = (GstBaseSrcClass *) klass;
218 gstaudiosrc_class = (GstAudioSrcClass *) klass;
220 gobject_class->finalize = gst_alsasrc_finalize;
221 gobject_class->get_property = gst_alsasrc_get_property;
222 gobject_class->set_property = gst_alsasrc_set_property;
224 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
226 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
227 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
228 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
229 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
230 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
231 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
232 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
233 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
235 g_object_class_install_property (gobject_class, PROP_DEVICE,
236 g_param_spec_string ("device", "Device",
237 "ALSA device, as defined in an asound configuration file",
238 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
240 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
241 g_param_spec_string ("device-name", "Device name",
242 "Human-readable name of the sound device",
243 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
245 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
246 g_param_spec_string ("card-name", "Card name",
247 "Human-readable name of the sound card",
248 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
250 g_object_class_install_property (gobject_class, PROP_IS_LIVE,
251 g_param_spec_boolean ("is-live",
254 0, G_PARAM_READWRITE));
257 static gboolean timer_flush_data(GstAlsaSrc *src)
259 if(src->handle_pause) {
262 frames = snd_pcm_avail(src->handle);
264 GST_DEBUG("Frames avail for reading: %d", frames);
265 err = snd_pcm_forward(src->handle,frames);
267 GST_WARNING("Can't recovery from suspend, prepare failed: %s", snd_strerror (err));
274 static GstStateChangeReturn
275 gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
277 GstAlsaSrc *src = GST_ALSA_SRC (element);
279 GstStateChangeReturn result = GST_STATE_CHANGE_FAILURE;
280 switch (transition) {
281 case GST_STATE_CHANGE_NULL_TO_READY:
282 GST_INFO("GST_STATE_CHANGE_NULL_TO_READY");
284 case GST_STATE_CHANGE_READY_TO_PAUSED:
285 GST_INFO("GST_STATE_CHANGE_READY_TO_PAUSED");
287 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
288 GST_INFO("GST_STATE_CHANGE_PAUSED_TO_PLAYING");
290 GstAudioSrc * asrc = (GstAudioSrc*)src;
292 GST_ALSA_SRC_LOCK (asrc);
295 frames = snd_pcm_avail(src->handle);
297 GST_DEBUG("Frames avail for reading: %d", frames);
298 err = snd_pcm_forward(src->handle,frames);
300 GST_WARNING("Can't recovery from suspend, prepare failed: %s", snd_strerror (err));
302 GST_ALSA_SRC_UNLOCK (asrc);
303 src->handle_pause = FALSE;
304 g_cond_signal(src->pause_cond);
307 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
308 GST_INFO("GST_STATE_CHANGE_PLAYING_TO_PAUSED");
310 src->handle_pause = TRUE;
311 src->flush_src_id = g_timeout_add(100, timer_flush_data, src);
314 case GST_STATE_CHANGE_PAUSED_TO_READY:
315 GST_INFO("GST_STATE_CHANGE_PAUSED_TO_READY");
317 if (src->flush_src_id > 0) g_source_remove(src->flush_src_id);
318 src->handle_pause = FALSE;
321 case GST_STATE_CHANGE_READY_TO_NULL:
322 GST_INFO("GST_STATE_CHANGE_READY_TO_NULL");
327 result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
332 gst_alsasrc_set_property (GObject * object, guint prop_id,
333 const GValue * value, GParamSpec * pspec)
337 src = GST_ALSA_SRC (object);
341 g_free (src->device);
342 src->device = g_value_dup_string (value);
343 if (src->device == NULL) {
344 src->device = g_strdup (DEFAULT_PROP_DEVICE);
348 src->is_live = g_value_get_boolean (value);
349 GST_DEBUG("is_live %d", src->is_live);
352 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
358 gst_alsasrc_get_property (GObject * object, guint prop_id,
359 GValue * value, GParamSpec * pspec)
363 src = GST_ALSA_SRC (object);
367 g_value_set_string (value, src->device);
369 case PROP_DEVICE_NAME:
370 g_value_take_string (value,
371 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
372 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
375 g_value_take_string (value,
376 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
377 src->device, SND_PCM_STREAM_CAPTURE));
380 g_value_set_boolean (value, src->is_live);
383 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
389 gst_alsasrc_init (GstAlsaSrc * alsasrc, GstAlsaSrcClass * g_class)
391 GST_DEBUG_OBJECT (alsasrc, "initializing");
393 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
394 alsasrc->cached_caps = NULL;
396 alsasrc->alsa_lock = g_mutex_new ();
397 alsasrc->pause_cond = g_cond_new ();
398 alsasrc->pause_lock = g_mutex_new ();
399 alsasrc->handle_pause = FALSE;
402 #define CHECK(call, error) \
404 if ((err = call) < 0) \
410 gst_alsasrc_getcaps (GstBaseSrc * bsrc)
412 GstElementClass *element_class;
413 GstPadTemplate *pad_template;
417 src = GST_ALSA_SRC (bsrc);
419 if (src->handle == NULL) {
420 GST_DEBUG_OBJECT (src, "device not open, using template caps");
421 return NULL; /* base class will get template caps for us */
424 if (src->cached_caps) {
425 GST_LOG_OBJECT (src, "Returning cached caps");
426 return gst_caps_ref (src->cached_caps);
429 element_class = GST_ELEMENT_GET_CLASS (src);
430 pad_template = gst_element_class_get_pad_template (element_class, "src");
431 g_return_val_if_fail (pad_template != NULL, NULL);
433 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
434 gst_pad_template_get_caps (pad_template));
437 src->cached_caps = gst_caps_ref (caps);
440 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
446 set_hwparams (GstAlsaSrc * alsa)
450 snd_pcm_hw_params_t *params;
452 snd_pcm_hw_params_malloc (¶ms);
454 /* choose all parameters */
455 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
456 /* set the interleaved read/write format */
457 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
459 /* set the sample format */
460 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
462 /* set the count of channels */
463 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
465 /* set the stream rate */
467 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
469 if (rrate != alsa->rate)
472 if (alsa->buffer_time != -1) {
473 /* set the buffer time */
474 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
475 &alsa->buffer_time, NULL), buffer_time);
477 if (alsa->period_time != -1) {
478 /* set the period time */
479 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
480 &alsa->period_time, NULL), period_time);
483 /* write the parameters to device */
484 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
486 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
489 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
492 snd_pcm_hw_params_free (params);
498 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
499 ("Broken configuration for recording: no configurations available: %s",
500 snd_strerror (err)));
501 snd_pcm_hw_params_free (params);
506 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
507 ("Access type not available for recording: %s", snd_strerror (err)));
508 snd_pcm_hw_params_free (params);
513 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
514 ("Sample format not available for recording: %s", snd_strerror (err)));
515 snd_pcm_hw_params_free (params);
522 if ((alsa->channels) == 1)
523 msg = g_strdup (_("Could not open device for recording in mono mode."));
524 if ((alsa->channels) == 2)
525 msg = g_strdup (_("Could not open device for recording in stereo mode."));
526 if ((alsa->channels) > 2)
529 ("Could not open device for recording in %d-channel mode"),
531 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
532 ("%s", snd_strerror (err)));
534 snd_pcm_hw_params_free (params);
539 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
540 ("Rate %iHz not available for recording: %s",
541 alsa->rate, snd_strerror (err)));
542 snd_pcm_hw_params_free (params);
547 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
548 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
549 snd_pcm_hw_params_free (params);
554 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
555 ("Unable to set buffer time %i for recording: %s",
556 alsa->buffer_time, snd_strerror (err)));
557 snd_pcm_hw_params_free (params);
562 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
563 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
564 snd_pcm_hw_params_free (params);
569 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
570 ("Unable to set period time %i for recording: %s", alsa->period_time,
571 snd_strerror (err)));
572 snd_pcm_hw_params_free (params);
577 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
578 ("Unable to get period size for recording: %s", snd_strerror (err)));
579 snd_pcm_hw_params_free (params);
584 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
585 ("Unable to set hw params for recording: %s", snd_strerror (err)));
586 snd_pcm_hw_params_free (params);
592 set_swparams (GstAlsaSrc * alsa)
595 snd_pcm_sw_params_t *params;
597 snd_pcm_sw_params_malloc (¶ms);
599 /* get the current swparams */
600 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
601 /* allow the transfer when at least period_size samples can be processed */
602 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
603 alsa->period_size), set_avail);
604 /* start the transfer on first read */
605 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
606 0), start_threshold);
608 #if GST_CHECK_ALSA_VERSION(1,0,16)
609 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
611 /* align all transfers to 1 sample */
612 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
615 /* write the parameters to the recording device */
616 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
618 snd_pcm_sw_params_free (params);
624 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
625 ("Unable to determine current swparams for playback: %s",
626 snd_strerror (err)));
627 snd_pcm_sw_params_free (params);
632 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
633 ("Unable to set start threshold mode for playback: %s",
634 snd_strerror (err)));
635 snd_pcm_sw_params_free (params);
640 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
641 ("Unable to set avail min for playback: %s", snd_strerror (err)));
642 snd_pcm_sw_params_free (params);
645 #if !GST_CHECK_ALSA_VERSION(1,0,16)
648 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
649 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
650 snd_pcm_sw_params_free (params);
656 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
657 ("Unable to set sw params for playback: %s", snd_strerror (err)));
658 snd_pcm_sw_params_free (params);
664 alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec)
666 switch (spec->type) {
667 case GST_BUFTYPE_LINEAR:
668 alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
669 spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
671 case GST_BUFTYPE_FLOAT:
672 switch (spec->format) {
674 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
677 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
680 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
683 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
689 case GST_BUFTYPE_A_LAW:
690 alsa->format = SND_PCM_FORMAT_A_LAW;
692 case GST_BUFTYPE_MU_LAW:
693 alsa->format = SND_PCM_FORMAT_MU_LAW;
699 alsa->rate = spec->rate;
700 alsa->channels = spec->channels;
701 alsa->buffer_time = spec->buffer_time;
702 alsa->period_time = spec->latency_time;
703 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
715 gst_alsasrc_open (GstAudioSrc * asrc)
720 alsa = GST_ALSA_SRC (asrc);
722 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
723 SND_PCM_NONBLOCK), open_error);
726 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
734 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
735 (_("Could not open audio device for recording. "
736 "Device is being used by another application.")),
737 ("Device '%s' is busy", alsa->device));
739 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
740 (_("Could not open audio device for recording.")),
741 ("Recording open error on device '%s': %s", alsa->device,
742 snd_strerror (err)));
749 gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
754 alsa = GST_ALSA_SRC (asrc);
756 if (!alsasrc_parse_spec (alsa, spec))
759 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
761 CHECK (set_hwparams (alsa), hw_params_failed);
762 CHECK (set_swparams (alsa), sw_params_failed);
763 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
765 alsa->bytes_per_sample = spec->bytes_per_sample;
766 spec->segsize = alsa->period_size * spec->bytes_per_sample;
767 spec->segtotal = alsa->buffer_size / alsa->period_size;
768 spec->silence_sample[0] = 0;
769 spec->silence_sample[1] = 0;
770 spec->silence_sample[2] = 0;
771 spec->silence_sample[3] = 0;
778 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
779 ("Error parsing spec"));
784 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
785 ("Could not set device to blocking: %s", snd_strerror (err)));
790 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
791 ("Setting of hwparams failed: %s", snd_strerror (err)));
796 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
797 ("Setting of swparams failed: %s", snd_strerror (err)));
802 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
803 ("Prepare failed: %s", snd_strerror (err)));
809 gst_alsasrc_unprepare (GstAudioSrc * asrc)
813 alsa = GST_ALSA_SRC (asrc);
815 snd_pcm_drop (alsa->handle);
816 snd_pcm_hw_free (alsa->handle);
817 snd_pcm_nonblock (alsa->handle, 1);
823 gst_alsasrc_close (GstAudioSrc * asrc)
825 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
827 snd_pcm_close (alsa->handle);
831 gst_alsa_mixer_free (alsa->mixer);
835 gst_caps_replace (&alsa->cached_caps, NULL);
841 * Underrun and suspend recovery
844 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
846 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
848 if (err == -EPIPE) { /* under-run */
849 GST_DEBUG_OBJECT (alsa, "overrun case");
850 err = snd_pcm_prepare (handle);
852 GST_WARNING_OBJECT (alsa,
853 "Can't recovery from underrun, prepare failed: %s",
856 } else if (err == -ESTRPIPE) {
857 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
858 g_usleep (100); /* wait until the suspend flag is released */
861 err = snd_pcm_prepare (handle);
863 GST_WARNING_OBJECT (alsa,
864 "Can't recovery from suspend, prepare failed: %s",
873 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
880 alsa = GST_ALSA_SRC (asrc);
882 cptr = length / alsa->bytes_per_sample;
885 if(alsa->is_live && alsa->handle_pause) g_cond_wait(alsa->pause_cond, alsa->pause_lock);
887 GST_ALSA_SRC_LOCK (asrc);
889 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
890 if (err == -EAGAIN) {
891 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
893 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
899 ptr += err * alsa->channels;
902 GST_ALSA_SRC_UNLOCK (asrc);
904 return length - (cptr * alsa->bytes_per_sample);
908 GST_ALSA_SRC_UNLOCK (asrc);
909 return length; /* skip one period */
914 gst_alsasrc_delay (GstAudioSrc * asrc)
917 snd_pcm_sframes_t delay;
920 alsa = GST_ALSA_SRC (asrc);
922 res = snd_pcm_delay (alsa->handle, &delay);
923 if (G_UNLIKELY (res < 0)) {
924 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
928 return CLAMP (delay, 0, alsa->buffer_size);
932 gst_alsasrc_reset (GstAudioSrc * asrc)
937 alsa = GST_ALSA_SRC (asrc);
939 GST_ALSA_SRC_LOCK (asrc);
940 GST_DEBUG_OBJECT (alsa, "drop");
941 CHECK (snd_pcm_drop (alsa->handle), drop_error);
942 GST_DEBUG_OBJECT (alsa, "prepare");
943 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
944 GST_DEBUG_OBJECT (alsa, "reset done");
945 GST_ALSA_SRC_UNLOCK (asrc);
952 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
954 GST_ALSA_SRC_UNLOCK (asrc);
959 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
961 GST_ALSA_SRC_UNLOCK (asrc);