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 GstRingBufferSpec * 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);
90 static GstStateChangeReturn gst_alsasrc_change_state (GstElement * element,
91 GstStateChange transition);
92 static GstFlowReturn gst_alsasrc_create (GstBaseSrc * bsrc, guint64 offset,
93 guint length, GstBuffer ** outbuf);
94 static GstClockTime gst_alsasrc_get_timestamp (GstAlsaSrc * src);
97 /* AlsaSrc signals and args */
103 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
104 # define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
106 # define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
109 static GstStaticPadTemplate alsasrc_src_factory =
110 GST_STATIC_PAD_TEMPLATE ("src",
113 GST_STATIC_CAPS ("audio/x-raw, "
114 "format = (string) " GST_AUDIO_FORMATS_ALL ", "
115 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
119 gst_alsasrc_finalize (GObject * object)
121 GstAlsaSrc *src = GST_ALSA_SRC (object);
123 g_free (src->device);
124 g_mutex_free (src->alsa_lock);
126 G_OBJECT_CLASS (parent_class)->finalize (object);
130 gst_alsasrc_init_interfaces (GType type)
132 static const GInterfaceInfo mixer_iface_info = {
133 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
138 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
140 gst_alsa_type_add_device_property_probe_interface (type);
144 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
146 GObjectClass *gobject_class;
147 GstElementClass *gstelement_class;
148 GstBaseSrcClass *gstbasesrc_class;
149 GstAudioSrcClass *gstaudiosrc_class;
151 gobject_class = (GObjectClass *) klass;
152 gstelement_class = (GstElementClass *) klass;
153 gstbasesrc_class = (GstBaseSrcClass *) klass;
154 gstaudiosrc_class = (GstAudioSrcClass *) klass;
156 gobject_class->finalize = gst_alsasrc_finalize;
157 gobject_class->get_property = gst_alsasrc_get_property;
158 gobject_class->set_property = gst_alsasrc_set_property;
160 gst_element_class_set_details_simple (gstelement_class,
161 "Audio source (ALSA)", "Source/Audio",
162 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
164 gst_element_class_add_pad_template (gstelement_class,
165 gst_static_pad_template_get (&alsasrc_src_factory));
167 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
169 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
170 gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_alsasrc_create);
172 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
173 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
174 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
175 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
176 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
177 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
178 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
180 g_object_class_install_property (gobject_class, PROP_DEVICE,
181 g_param_spec_string ("device", "Device",
182 "ALSA device, as defined in an asound configuration file",
183 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
185 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
186 g_param_spec_string ("device-name", "Device name",
187 "Human-readable name of the sound device",
188 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
190 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
191 g_param_spec_string ("card-name", "Card name",
192 "Human-readable name of the sound card",
193 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
197 gst_alsasrc_get_timestamp (GstAlsaSrc * src)
199 snd_pcm_status_t *status;
200 snd_htimestamp_t tstamp;
201 GstClockTime timestamp;
202 snd_pcm_uframes_t availmax;
204 GST_DEBUG_OBJECT (src, "Getting alsa timestamp!");
207 GST_ERROR_OBJECT (src, "No alsa handle created yet !");
211 if (snd_pcm_status_malloc (&status) != 0) {
212 GST_ERROR_OBJECT (src, "snd_pcm_status_malloc failed");
215 if (snd_pcm_status (src->handle, status) != 0) {
216 GST_ERROR_OBJECT (src, "snd_pcm_status failed");
219 /* get high resolution time stamp from driver */
220 snd_pcm_status_get_htstamp (status, &tstamp);
221 timestamp = GST_TIMESPEC_TO_TIME (tstamp);
223 /* Max available frames sets the depth of the buffer */
224 availmax = snd_pcm_status_get_avail_max (status);
226 /* Compensate the fact that the timestamp references the last sample */
227 timestamp -= gst_util_uint64_scale_int (availmax * 2, GST_SECOND, src->rate);
228 /* Compensate for the delay until the package is available */
229 timestamp += gst_util_uint64_scale_int (snd_pcm_status_get_delay (status),
230 GST_SECOND, src->rate);
232 snd_pcm_status_free (status);
234 GST_DEBUG_OBJECT (src, "ALSA timestamp : %" GST_TIME_FORMAT,
235 GST_TIME_ARGS (timestamp));
240 gst_alsasrc_set_property (GObject * object, guint prop_id,
241 const GValue * value, GParamSpec * pspec)
245 src = GST_ALSA_SRC (object);
249 g_free (src->device);
250 src->device = g_value_dup_string (value);
251 if (src->device == NULL) {
252 src->device = g_strdup (DEFAULT_PROP_DEVICE);
256 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
262 gst_alsasrc_get_property (GObject * object, guint prop_id,
263 GValue * value, GParamSpec * pspec)
267 src = GST_ALSA_SRC (object);
271 g_value_set_string (value, src->device);
273 case PROP_DEVICE_NAME:
274 g_value_take_string (value,
275 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
276 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
279 g_value_take_string (value,
280 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
281 src->device, SND_PCM_STREAM_CAPTURE));
284 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
289 static GstStateChangeReturn
290 gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
292 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
293 GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
294 GstAlsaSrc *asrc = GST_ALSA_SRC (element);
297 switch (transition) {
298 /* Show the compiler that we care */
299 case GST_STATE_CHANGE_NULL_TO_READY:
300 case GST_STATE_CHANGE_READY_TO_PAUSED:
301 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
302 case GST_STATE_CHANGE_PAUSED_TO_READY:
303 case GST_STATE_CHANGE_READY_TO_NULL:
306 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
308 asrc->driver_timestamps = FALSE;
309 if (GST_IS_SYSTEM_CLOCK (clk)) {
311 g_object_get (clk, "clock-type", &clocktype, NULL);
312 if (clocktype == GST_CLOCK_TYPE_MONOTONIC) {
313 asrc->driver_timestamps = TRUE;
318 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
324 gst_alsasrc_create (GstBaseSrc * bsrc, guint64 offset, guint length,
327 GstFlowReturn ret = GST_FLOW_OK;
328 GstAlsaSrc *asrc = GST_ALSA_SRC (bsrc);
331 GST_BASE_SRC_CLASS (parent_class)->create (bsrc, offset, length, outbuf);
332 if (asrc->driver_timestamps == TRUE && *outbuf) {
333 GST_BUFFER_TIMESTAMP (*outbuf) =
334 gst_alsasrc_get_timestamp ((GstAlsaSrc *) bsrc);
341 gst_alsasrc_init (GstAlsaSrc * alsasrc)
343 GST_DEBUG_OBJECT (alsasrc, "initializing");
345 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
346 alsasrc->cached_caps = NULL;
347 alsasrc->driver_timestamps = FALSE;
349 alsasrc->alsa_lock = g_mutex_new ();
352 #define CHECK(call, error) \
354 if ((err = call) < 0) \
360 gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
362 GstElementClass *element_class;
363 GstPadTemplate *pad_template;
365 GstCaps *caps, *templ_caps;
367 src = GST_ALSA_SRC (bsrc);
369 if (src->handle == NULL) {
370 GST_DEBUG_OBJECT (src, "device not open, using template caps");
371 return NULL; /* base class will get template caps for us */
374 if (src->cached_caps) {
375 GST_LOG_OBJECT (src, "Returning cached caps");
377 return gst_caps_intersect_full (filter, src->cached_caps,
378 GST_CAPS_INTERSECT_FIRST);
380 return gst_caps_ref (src->cached_caps);
383 element_class = GST_ELEMENT_GET_CLASS (src);
384 pad_template = gst_element_class_get_pad_template (element_class, "src");
385 g_return_val_if_fail (pad_template != NULL, NULL);
387 templ_caps = gst_pad_template_get_caps (pad_template);
388 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
390 gst_caps_unref (templ_caps);
393 src->cached_caps = gst_caps_ref (caps);
396 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
399 GstCaps *intersection;
402 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
403 gst_caps_unref (caps);
411 set_hwparams (GstAlsaSrc * alsa)
415 snd_pcm_hw_params_t *params;
417 snd_pcm_hw_params_malloc (¶ms);
419 /* choose all parameters */
420 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
421 /* set the interleaved read/write format */
422 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
424 /* set the sample format */
425 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
427 /* set the count of channels */
428 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
430 /* set the stream rate */
432 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
434 if (rrate != alsa->rate)
437 if (alsa->buffer_time != -1) {
438 /* set the buffer time */
439 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
440 &alsa->buffer_time, NULL), buffer_time);
442 if (alsa->period_time != -1) {
443 /* set the period time */
444 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
445 &alsa->period_time, NULL), period_time);
448 /* write the parameters to device */
449 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
451 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
454 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
457 snd_pcm_hw_params_free (params);
463 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
464 ("Broken configuration for recording: no configurations available: %s",
465 snd_strerror (err)));
466 snd_pcm_hw_params_free (params);
471 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
472 ("Access type not available for recording: %s", snd_strerror (err)));
473 snd_pcm_hw_params_free (params);
478 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
479 ("Sample format not available for recording: %s", snd_strerror (err)));
480 snd_pcm_hw_params_free (params);
487 if ((alsa->channels) == 1)
488 msg = g_strdup (_("Could not open device for recording in mono mode."));
489 if ((alsa->channels) == 2)
490 msg = g_strdup (_("Could not open device for recording in stereo mode."));
491 if ((alsa->channels) > 2)
494 ("Could not open device for recording in %d-channel mode"),
496 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
497 ("%s", snd_strerror (err)));
499 snd_pcm_hw_params_free (params);
504 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
505 ("Rate %iHz not available for recording: %s",
506 alsa->rate, snd_strerror (err)));
507 snd_pcm_hw_params_free (params);
512 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
513 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
514 snd_pcm_hw_params_free (params);
519 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
520 ("Unable to set buffer time %i for recording: %s",
521 alsa->buffer_time, snd_strerror (err)));
522 snd_pcm_hw_params_free (params);
527 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
528 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
529 snd_pcm_hw_params_free (params);
534 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
535 ("Unable to set period time %i for recording: %s", alsa->period_time,
536 snd_strerror (err)));
537 snd_pcm_hw_params_free (params);
542 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
543 ("Unable to get period size for recording: %s", snd_strerror (err)));
544 snd_pcm_hw_params_free (params);
549 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
550 ("Unable to set hw params for recording: %s", snd_strerror (err)));
551 snd_pcm_hw_params_free (params);
557 set_swparams (GstAlsaSrc * alsa)
560 snd_pcm_sw_params_t *params;
562 snd_pcm_sw_params_malloc (¶ms);
564 /* get the current swparams */
565 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
566 /* allow the transfer when at least period_size samples can be processed */
567 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
568 alsa->period_size), set_avail);
569 /* start the transfer on first read */
570 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
571 0), start_threshold);
573 #if GST_CHECK_ALSA_VERSION(1,0,16)
574 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
576 /* align all transfers to 1 sample */
577 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
580 /* write the parameters to the recording device */
581 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
583 snd_pcm_sw_params_free (params);
589 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
590 ("Unable to determine current swparams for playback: %s",
591 snd_strerror (err)));
592 snd_pcm_sw_params_free (params);
597 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
598 ("Unable to set start threshold mode for playback: %s",
599 snd_strerror (err)));
600 snd_pcm_sw_params_free (params);
605 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
606 ("Unable to set avail min for playback: %s", snd_strerror (err)));
607 snd_pcm_sw_params_free (params);
610 #if !GST_CHECK_ALSA_VERSION(1,0,16)
613 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
614 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
615 snd_pcm_sw_params_free (params);
621 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
622 ("Unable to set sw params for playback: %s", snd_strerror (err)));
623 snd_pcm_sw_params_free (params);
629 alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec)
631 switch (spec->type) {
632 case GST_BUFTYPE_RAW:
633 switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
634 case GST_AUDIO_FORMAT_U8:
635 alsa->format = SND_PCM_FORMAT_U8;
637 case GST_AUDIO_FORMAT_S8:
638 alsa->format = SND_PCM_FORMAT_S8;
640 case GST_AUDIO_FORMAT_S16LE:
641 alsa->format = SND_PCM_FORMAT_S16_LE;
643 case GST_AUDIO_FORMAT_S16BE:
644 alsa->format = SND_PCM_FORMAT_S16_BE;
646 case GST_AUDIO_FORMAT_U16LE:
647 alsa->format = SND_PCM_FORMAT_U16_LE;
649 case GST_AUDIO_FORMAT_U16BE:
650 alsa->format = SND_PCM_FORMAT_U16_BE;
652 case GST_AUDIO_FORMAT_S24_32LE:
653 alsa->format = SND_PCM_FORMAT_S24_LE;
655 case GST_AUDIO_FORMAT_S24_32BE:
656 alsa->format = SND_PCM_FORMAT_S24_BE;
658 case GST_AUDIO_FORMAT_U24_32LE:
659 alsa->format = SND_PCM_FORMAT_U24_LE;
661 case GST_AUDIO_FORMAT_U24_32BE:
662 alsa->format = SND_PCM_FORMAT_U24_BE;
664 case GST_AUDIO_FORMAT_S32LE:
665 alsa->format = SND_PCM_FORMAT_S32_LE;
667 case GST_AUDIO_FORMAT_S32BE:
668 alsa->format = SND_PCM_FORMAT_S32_BE;
670 case GST_AUDIO_FORMAT_U32LE:
671 alsa->format = SND_PCM_FORMAT_U32_LE;
673 case GST_AUDIO_FORMAT_U32BE:
674 alsa->format = SND_PCM_FORMAT_U32_BE;
676 case GST_AUDIO_FORMAT_S24LE:
677 alsa->format = SND_PCM_FORMAT_S24_3LE;
679 case GST_AUDIO_FORMAT_S24BE:
680 alsa->format = SND_PCM_FORMAT_S24_3BE;
682 case GST_AUDIO_FORMAT_U24LE:
683 alsa->format = SND_PCM_FORMAT_U24_3LE;
685 case GST_AUDIO_FORMAT_U24BE:
686 alsa->format = SND_PCM_FORMAT_U24_3BE;
688 case GST_AUDIO_FORMAT_S20LE:
689 alsa->format = SND_PCM_FORMAT_S20_3LE;
691 case GST_AUDIO_FORMAT_S20BE:
692 alsa->format = SND_PCM_FORMAT_S20_3BE;
694 case GST_AUDIO_FORMAT_U20LE:
695 alsa->format = SND_PCM_FORMAT_U20_3LE;
697 case GST_AUDIO_FORMAT_U20BE:
698 alsa->format = SND_PCM_FORMAT_U20_3BE;
700 case GST_AUDIO_FORMAT_S18LE:
701 alsa->format = SND_PCM_FORMAT_S18_3LE;
703 case GST_AUDIO_FORMAT_S18BE:
704 alsa->format = SND_PCM_FORMAT_S18_3BE;
706 case GST_AUDIO_FORMAT_U18LE:
707 alsa->format = SND_PCM_FORMAT_U18_3LE;
709 case GST_AUDIO_FORMAT_U18BE:
710 alsa->format = SND_PCM_FORMAT_U18_3BE;
712 case GST_AUDIO_FORMAT_F32LE:
713 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
715 case GST_AUDIO_FORMAT_F32BE:
716 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
718 case GST_AUDIO_FORMAT_F64LE:
719 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
721 case GST_AUDIO_FORMAT_F64BE:
722 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
728 case GST_BUFTYPE_A_LAW:
729 alsa->format = SND_PCM_FORMAT_A_LAW;
731 case GST_BUFTYPE_MU_LAW:
732 alsa->format = SND_PCM_FORMAT_MU_LAW;
738 alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
739 alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
740 alsa->buffer_time = spec->buffer_time;
741 alsa->period_time = spec->latency_time;
742 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
754 gst_alsasrc_open (GstAudioSrc * asrc)
759 alsa = GST_ALSA_SRC (asrc);
761 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
762 SND_PCM_NONBLOCK), open_error);
765 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
773 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
774 (_("Could not open audio device for recording. "
775 "Device is being used by another application.")),
776 ("Device '%s' is busy", alsa->device));
778 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
779 (_("Could not open audio device for recording.")),
780 ("Recording open error on device '%s': %s", alsa->device,
781 snd_strerror (err)));
788 gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
793 alsa = GST_ALSA_SRC (asrc);
795 if (!alsasrc_parse_spec (alsa, spec))
798 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
800 CHECK (set_hwparams (alsa), hw_params_failed);
801 CHECK (set_swparams (alsa), sw_params_failed);
802 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
804 alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
805 spec->segsize = alsa->period_size * alsa->bpf;
806 spec->segtotal = alsa->buffer_size / alsa->period_size;
813 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
814 ("Error parsing spec"));
819 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
820 ("Could not set device to blocking: %s", snd_strerror (err)));
825 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
826 ("Setting of hwparams failed: %s", snd_strerror (err)));
831 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
832 ("Setting of swparams failed: %s", snd_strerror (err)));
837 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
838 ("Prepare failed: %s", snd_strerror (err)));
844 gst_alsasrc_unprepare (GstAudioSrc * asrc)
848 alsa = GST_ALSA_SRC (asrc);
850 snd_pcm_drop (alsa->handle);
851 snd_pcm_hw_free (alsa->handle);
852 snd_pcm_nonblock (alsa->handle, 1);
858 gst_alsasrc_close (GstAudioSrc * asrc)
860 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
862 snd_pcm_close (alsa->handle);
866 gst_alsa_mixer_free (alsa->mixer);
870 gst_caps_replace (&alsa->cached_caps, NULL);
876 * Underrun and suspend recovery
879 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
881 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
883 if (err == -EPIPE) { /* under-run */
884 err = snd_pcm_prepare (handle);
886 GST_WARNING_OBJECT (alsa,
887 "Can't recovery from underrun, prepare failed: %s",
890 } else if (err == -ESTRPIPE) {
891 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
892 g_usleep (100); /* wait until the suspend flag is released */
895 err = snd_pcm_prepare (handle);
897 GST_WARNING_OBJECT (alsa,
898 "Can't recovery from suspend, prepare failed: %s",
907 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
914 alsa = GST_ALSA_SRC (asrc);
916 cptr = length / alsa->bpf;
919 GST_ALSA_SRC_LOCK (asrc);
921 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
922 if (err == -EAGAIN) {
923 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
925 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
931 ptr += err * alsa->channels;
934 GST_ALSA_SRC_UNLOCK (asrc);
936 return length - (cptr * alsa->bpf);
940 GST_ALSA_SRC_UNLOCK (asrc);
941 return length; /* skip one period */
946 gst_alsasrc_delay (GstAudioSrc * asrc)
949 snd_pcm_sframes_t delay;
952 alsa = GST_ALSA_SRC (asrc);
954 res = snd_pcm_delay (alsa->handle, &delay);
955 if (G_UNLIKELY (res < 0)) {
956 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
960 return CLAMP (delay, 0, alsa->buffer_size);
964 gst_alsasrc_reset (GstAudioSrc * asrc)
969 alsa = GST_ALSA_SRC (asrc);
971 GST_ALSA_SRC_LOCK (asrc);
972 GST_DEBUG_OBJECT (alsa, "drop");
973 CHECK (snd_pcm_drop (alsa->handle), drop_error);
974 GST_DEBUG_OBJECT (alsa, "prepare");
975 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
976 GST_DEBUG_OBJECT (alsa, "reset done");
977 GST_ALSA_SRC_UNLOCK (asrc);
984 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
986 GST_ALSA_SRC_UNLOCK (asrc);
991 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
993 GST_ALSA_SRC_UNLOCK (asrc);