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);
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;
205 GST_DEBUG_OBJECT (src, "Getting alsa timestamp!");
208 GST_ERROR_OBJECT (src, "No alsa handle created yet !");
209 return GST_CLOCK_TIME_NONE;
212 if (snd_pcm_status_malloc (&status) != 0) {
213 GST_ERROR_OBJECT (src, "snd_pcm_status_malloc failed");
214 return GST_CLOCK_TIME_NONE;
217 if (snd_pcm_status (src->handle, status) != 0) {
218 GST_ERROR_OBJECT (src, "snd_pcm_status failed");
219 snd_pcm_status_free (status);
220 return GST_CLOCK_TIME_NONE;
223 /* get high resolution time stamp from driver */
224 snd_pcm_status_get_htstamp (status, &tstamp);
225 timestamp = GST_TIMESPEC_TO_TIME (tstamp);
226 GST_DEBUG_OBJECT (src, "Base ts: %" GST_TIME_FORMAT,
227 GST_TIME_ARGS (timestamp));
228 if (timestamp == 0) {
229 /* This timestamp is supposed to represent the last sample, so 0 (which
230 can be returned on some ALSA setups (such as mine)) must mean that it
231 is invalid, unless there's just one sample, but we'll ignore that. */
232 GST_WARNING_OBJECT (src,
233 "No timestamp returned from snd_pcm_status_get_htstamp");
234 return GST_CLOCK_TIME_NONE;
237 /* Max available frames sets the depth of the buffer */
238 availmax = snd_pcm_status_get_avail_max (status);
240 /* Compensate the fact that the timestamp references the last sample */
241 offset = -gst_util_uint64_scale_int (availmax * 2, GST_SECOND, src->rate);
242 /* Compensate for the delay until the package is available */
243 offset += gst_util_uint64_scale_int (snd_pcm_status_get_delay (status),
244 GST_SECOND, src->rate);
246 snd_pcm_status_free (status);
248 /* just in case, should not happen */
249 if (-offset > timestamp)
254 /* Take first ts into account */
255 if (src->first_alsa_ts == GST_CLOCK_TIME_NONE) {
256 src->first_alsa_ts = timestamp;
258 timestamp -= src->first_alsa_ts;
260 GST_DEBUG_OBJECT (src, "ALSA timestamp : %" GST_TIME_FORMAT,
261 GST_TIME_ARGS (timestamp));
266 gst_alsasrc_set_property (GObject * object, guint prop_id,
267 const GValue * value, GParamSpec * pspec)
271 src = GST_ALSA_SRC (object);
275 g_free (src->device);
276 src->device = g_value_dup_string (value);
277 if (src->device == NULL) {
278 src->device = g_strdup (DEFAULT_PROP_DEVICE);
282 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
288 gst_alsasrc_get_property (GObject * object, guint prop_id,
289 GValue * value, GParamSpec * pspec)
293 src = GST_ALSA_SRC (object);
297 g_value_set_string (value, src->device);
299 case PROP_DEVICE_NAME:
300 g_value_take_string (value,
301 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
302 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
305 g_value_take_string (value,
306 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
307 src->device, SND_PCM_STREAM_CAPTURE));
310 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
315 static GstStateChangeReturn
316 gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
318 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
319 GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (element);
320 GstAlsaSrc *asrc = GST_ALSA_SRC (element);
323 switch (transition) {
324 /* Show the compiler that we care */
325 case GST_STATE_CHANGE_NULL_TO_READY:
326 case GST_STATE_CHANGE_READY_TO_PAUSED:
327 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
328 case GST_STATE_CHANGE_PAUSED_TO_READY:
329 case GST_STATE_CHANGE_READY_TO_NULL:
332 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
334 asrc->driver_timestamps = FALSE;
335 if (GST_IS_SYSTEM_CLOCK (clk)) {
337 g_object_get (clk, "clock-type", &clocktype, NULL);
338 if (clocktype == GST_CLOCK_TYPE_MONOTONIC) {
339 asrc->driver_timestamps = TRUE;
344 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
350 gst_alsasrc_create (GstBaseSrc * bsrc, guint64 offset, guint length,
353 GstFlowReturn ret = GST_FLOW_OK;
354 GstAlsaSrc *asrc = GST_ALSA_SRC (bsrc);
357 GST_BASE_SRC_CLASS (parent_class)->create (bsrc, offset, length, outbuf);
358 if (asrc->driver_timestamps == TRUE && *outbuf) {
359 GstClockTime ts = gst_alsasrc_get_timestamp (asrc);
360 if (GST_CLOCK_TIME_IS_VALID (ts)) {
361 GST_BUFFER_TIMESTAMP (*outbuf) = ts;
369 gst_alsasrc_init (GstAlsaSrc * alsasrc)
371 GST_DEBUG_OBJECT (alsasrc, "initializing");
373 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
374 alsasrc->cached_caps = NULL;
375 alsasrc->driver_timestamps = FALSE;
376 alsasrc->first_alsa_ts = GST_CLOCK_TIME_NONE;
378 alsasrc->alsa_lock = g_mutex_new ();
381 #define CHECK(call, error) \
383 if ((err = call) < 0) \
389 gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
391 GstElementClass *element_class;
392 GstPadTemplate *pad_template;
394 GstCaps *caps, *templ_caps;
396 src = GST_ALSA_SRC (bsrc);
398 if (src->handle == NULL) {
399 GST_DEBUG_OBJECT (src, "device not open, using template caps");
400 return GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
403 if (src->cached_caps) {
404 GST_LOG_OBJECT (src, "Returning cached caps");
406 return gst_caps_intersect_full (filter, src->cached_caps,
407 GST_CAPS_INTERSECT_FIRST);
409 return gst_caps_ref (src->cached_caps);
412 element_class = GST_ELEMENT_GET_CLASS (src);
413 pad_template = gst_element_class_get_pad_template (element_class, "src");
414 g_return_val_if_fail (pad_template != NULL, NULL);
416 templ_caps = gst_pad_template_get_caps (pad_template);
417 GST_INFO_OBJECT (src, "template caps %" GST_PTR_FORMAT, templ_caps);
419 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
421 gst_caps_unref (templ_caps);
424 src->cached_caps = gst_caps_ref (caps);
427 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
430 GstCaps *intersection;
433 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
434 gst_caps_unref (caps);
442 set_hwparams (GstAlsaSrc * alsa)
446 snd_pcm_hw_params_t *params;
448 snd_pcm_hw_params_malloc (¶ms);
450 /* choose all parameters */
451 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
452 /* set the interleaved read/write format */
453 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
455 /* set the sample format */
456 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
458 /* set the count of channels */
459 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
461 /* set the stream rate */
463 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
465 if (rrate != alsa->rate)
468 if (alsa->buffer_time != -1) {
469 /* set the buffer time */
470 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
471 &alsa->buffer_time, NULL), buffer_time);
473 if (alsa->period_time != -1) {
474 /* set the period time */
475 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
476 &alsa->period_time, NULL), period_time);
479 /* write the parameters to device */
480 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
482 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
485 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
488 snd_pcm_hw_params_free (params);
494 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
495 ("Broken configuration for recording: no configurations available: %s",
496 snd_strerror (err)));
497 snd_pcm_hw_params_free (params);
502 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
503 ("Access type not available for recording: %s", snd_strerror (err)));
504 snd_pcm_hw_params_free (params);
509 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
510 ("Sample format not available for recording: %s", snd_strerror (err)));
511 snd_pcm_hw_params_free (params);
518 if ((alsa->channels) == 1)
519 msg = g_strdup (_("Could not open device for recording in mono mode."));
520 if ((alsa->channels) == 2)
521 msg = g_strdup (_("Could not open device for recording in stereo mode."));
522 if ((alsa->channels) > 2)
525 ("Could not open device for recording in %d-channel mode"),
527 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
528 ("%s", snd_strerror (err)));
530 snd_pcm_hw_params_free (params);
535 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
536 ("Rate %iHz not available for recording: %s",
537 alsa->rate, snd_strerror (err)));
538 snd_pcm_hw_params_free (params);
543 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
544 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
545 snd_pcm_hw_params_free (params);
550 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
551 ("Unable to set buffer time %i for recording: %s",
552 alsa->buffer_time, snd_strerror (err)));
553 snd_pcm_hw_params_free (params);
558 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
559 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
560 snd_pcm_hw_params_free (params);
565 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
566 ("Unable to set period time %i for recording: %s", alsa->period_time,
567 snd_strerror (err)));
568 snd_pcm_hw_params_free (params);
573 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
574 ("Unable to get period size for recording: %s", snd_strerror (err)));
575 snd_pcm_hw_params_free (params);
580 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
581 ("Unable to set hw params for recording: %s", snd_strerror (err)));
582 snd_pcm_hw_params_free (params);
588 set_swparams (GstAlsaSrc * alsa)
591 snd_pcm_sw_params_t *params;
593 snd_pcm_sw_params_malloc (¶ms);
595 /* get the current swparams */
596 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
597 /* allow the transfer when at least period_size samples can be processed */
598 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
599 alsa->period_size), set_avail);
600 /* start the transfer on first read */
601 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
602 0), start_threshold);
604 #if GST_CHECK_ALSA_VERSION(1,0,16)
605 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
607 /* align all transfers to 1 sample */
608 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
611 /* write the parameters to the recording device */
612 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
614 snd_pcm_sw_params_free (params);
620 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
621 ("Unable to determine current swparams for playback: %s",
622 snd_strerror (err)));
623 snd_pcm_sw_params_free (params);
628 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
629 ("Unable to set start threshold mode for playback: %s",
630 snd_strerror (err)));
631 snd_pcm_sw_params_free (params);
636 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
637 ("Unable to set avail min for playback: %s", snd_strerror (err)));
638 snd_pcm_sw_params_free (params);
641 #if !GST_CHECK_ALSA_VERSION(1,0,16)
644 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
645 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
646 snd_pcm_sw_params_free (params);
652 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
653 ("Unable to set sw params for playback: %s", snd_strerror (err)));
654 snd_pcm_sw_params_free (params);
660 alsasrc_parse_spec (GstAlsaSrc * alsa, GstAudioRingBufferSpec * spec)
662 switch (spec->type) {
663 case GST_BUFTYPE_RAW:
664 switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
665 case GST_AUDIO_FORMAT_U8:
666 alsa->format = SND_PCM_FORMAT_U8;
668 case GST_AUDIO_FORMAT_S8:
669 alsa->format = SND_PCM_FORMAT_S8;
671 case GST_AUDIO_FORMAT_S16LE:
672 alsa->format = SND_PCM_FORMAT_S16_LE;
674 case GST_AUDIO_FORMAT_S16BE:
675 alsa->format = SND_PCM_FORMAT_S16_BE;
677 case GST_AUDIO_FORMAT_U16LE:
678 alsa->format = SND_PCM_FORMAT_U16_LE;
680 case GST_AUDIO_FORMAT_U16BE:
681 alsa->format = SND_PCM_FORMAT_U16_BE;
683 case GST_AUDIO_FORMAT_S24_32LE:
684 alsa->format = SND_PCM_FORMAT_S24_LE;
686 case GST_AUDIO_FORMAT_S24_32BE:
687 alsa->format = SND_PCM_FORMAT_S24_BE;
689 case GST_AUDIO_FORMAT_U24_32LE:
690 alsa->format = SND_PCM_FORMAT_U24_LE;
692 case GST_AUDIO_FORMAT_U24_32BE:
693 alsa->format = SND_PCM_FORMAT_U24_BE;
695 case GST_AUDIO_FORMAT_S32LE:
696 alsa->format = SND_PCM_FORMAT_S32_LE;
698 case GST_AUDIO_FORMAT_S32BE:
699 alsa->format = SND_PCM_FORMAT_S32_BE;
701 case GST_AUDIO_FORMAT_U32LE:
702 alsa->format = SND_PCM_FORMAT_U32_LE;
704 case GST_AUDIO_FORMAT_U32BE:
705 alsa->format = SND_PCM_FORMAT_U32_BE;
707 case GST_AUDIO_FORMAT_S24LE:
708 alsa->format = SND_PCM_FORMAT_S24_3LE;
710 case GST_AUDIO_FORMAT_S24BE:
711 alsa->format = SND_PCM_FORMAT_S24_3BE;
713 case GST_AUDIO_FORMAT_U24LE:
714 alsa->format = SND_PCM_FORMAT_U24_3LE;
716 case GST_AUDIO_FORMAT_U24BE:
717 alsa->format = SND_PCM_FORMAT_U24_3BE;
719 case GST_AUDIO_FORMAT_S20LE:
720 alsa->format = SND_PCM_FORMAT_S20_3LE;
722 case GST_AUDIO_FORMAT_S20BE:
723 alsa->format = SND_PCM_FORMAT_S20_3BE;
725 case GST_AUDIO_FORMAT_U20LE:
726 alsa->format = SND_PCM_FORMAT_U20_3LE;
728 case GST_AUDIO_FORMAT_U20BE:
729 alsa->format = SND_PCM_FORMAT_U20_3BE;
731 case GST_AUDIO_FORMAT_S18LE:
732 alsa->format = SND_PCM_FORMAT_S18_3LE;
734 case GST_AUDIO_FORMAT_S18BE:
735 alsa->format = SND_PCM_FORMAT_S18_3BE;
737 case GST_AUDIO_FORMAT_U18LE:
738 alsa->format = SND_PCM_FORMAT_U18_3LE;
740 case GST_AUDIO_FORMAT_U18BE:
741 alsa->format = SND_PCM_FORMAT_U18_3BE;
743 case GST_AUDIO_FORMAT_F32LE:
744 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
746 case GST_AUDIO_FORMAT_F32BE:
747 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
749 case GST_AUDIO_FORMAT_F64LE:
750 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
752 case GST_AUDIO_FORMAT_F64BE:
753 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
759 case GST_BUFTYPE_A_LAW:
760 alsa->format = SND_PCM_FORMAT_A_LAW;
762 case GST_BUFTYPE_MU_LAW:
763 alsa->format = SND_PCM_FORMAT_MU_LAW;
769 alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
770 alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
771 alsa->buffer_time = spec->buffer_time;
772 alsa->period_time = spec->latency_time;
773 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
785 gst_alsasrc_open (GstAudioSrc * asrc)
790 alsa = GST_ALSA_SRC (asrc);
792 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
793 SND_PCM_NONBLOCK), open_error);
796 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
804 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
805 (_("Could not open audio device for recording. "
806 "Device is being used by another application.")),
807 ("Device '%s' is busy", alsa->device));
809 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
810 (_("Could not open audio device for recording.")),
811 ("Recording open error on device '%s': %s", alsa->device,
812 snd_strerror (err)));
819 gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
824 alsa = GST_ALSA_SRC (asrc);
826 if (!alsasrc_parse_spec (alsa, spec))
829 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
831 CHECK (set_hwparams (alsa), hw_params_failed);
832 CHECK (set_swparams (alsa), sw_params_failed);
833 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
835 alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
836 spec->segsize = alsa->period_size * alsa->bpf;
837 spec->segtotal = alsa->buffer_size / alsa->period_size;
844 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
845 ("Error parsing spec"));
850 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
851 ("Could not set device to blocking: %s", snd_strerror (err)));
856 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
857 ("Setting of hwparams failed: %s", snd_strerror (err)));
862 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
863 ("Setting of swparams failed: %s", snd_strerror (err)));
868 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
869 ("Prepare failed: %s", snd_strerror (err)));
875 gst_alsasrc_unprepare (GstAudioSrc * asrc)
879 alsa = GST_ALSA_SRC (asrc);
881 snd_pcm_drop (alsa->handle);
882 snd_pcm_hw_free (alsa->handle);
883 snd_pcm_nonblock (alsa->handle, 1);
889 gst_alsasrc_close (GstAudioSrc * asrc)
891 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
893 snd_pcm_close (alsa->handle);
897 gst_alsa_mixer_free (alsa->mixer);
901 gst_caps_replace (&alsa->cached_caps, NULL);
907 * Underrun and suspend recovery
910 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
912 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
914 if (err == -EPIPE) { /* under-run */
915 err = snd_pcm_prepare (handle);
917 GST_WARNING_OBJECT (alsa,
918 "Can't recovery from underrun, prepare failed: %s",
921 } else if (err == -ESTRPIPE) {
922 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
923 g_usleep (100); /* wait until the suspend flag is released */
926 err = snd_pcm_prepare (handle);
928 GST_WARNING_OBJECT (alsa,
929 "Can't recovery from suspend, prepare failed: %s",
938 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
945 alsa = GST_ALSA_SRC (asrc);
947 cptr = length / alsa->bpf;
950 GST_ALSA_SRC_LOCK (asrc);
952 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
953 if (err == -EAGAIN) {
954 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
956 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
962 ptr += err * alsa->channels;
965 GST_ALSA_SRC_UNLOCK (asrc);
967 return length - (cptr * alsa->bpf);
971 GST_ALSA_SRC_UNLOCK (asrc);
972 return length; /* skip one period */
977 gst_alsasrc_delay (GstAudioSrc * asrc)
980 snd_pcm_sframes_t delay;
983 alsa = GST_ALSA_SRC (asrc);
985 res = snd_pcm_delay (alsa->handle, &delay);
986 if (G_UNLIKELY (res < 0)) {
987 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
991 return CLAMP (delay, 0, alsa->buffer_size);
995 gst_alsasrc_reset (GstAudioSrc * asrc)
1000 alsa = GST_ALSA_SRC (asrc);
1002 GST_ALSA_SRC_LOCK (asrc);
1003 GST_DEBUG_OBJECT (alsa, "drop");
1004 CHECK (snd_pcm_drop (alsa->handle), drop_error);
1005 GST_DEBUG_OBJECT (alsa, "prepare");
1006 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
1007 GST_DEBUG_OBJECT (alsa, "reset done");
1008 alsa->first_alsa_ts = GST_CLOCK_TIME_NONE;
1009 GST_ALSA_SRC_UNLOCK (asrc);
1016 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
1017 snd_strerror (err));
1018 GST_ALSA_SRC_UNLOCK (asrc);
1023 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
1024 snd_strerror (err));
1025 GST_ALSA_SRC_UNLOCK (asrc);