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);
69 GST_BOILERPLATE_FULL (GstAlsaSrc, gst_alsasrc, GstAudioSrc,
70 GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces);
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);
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-int, "
114 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
115 "signed = (boolean) { TRUE, FALSE }, "
118 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
120 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
121 "signed = (boolean) { TRUE, FALSE }, "
124 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
126 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
127 "signed = (boolean) { TRUE, FALSE }, "
130 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
132 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, "
133 "signed = (boolean) { TRUE, FALSE }, "
136 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
138 "signed = (boolean) { TRUE, FALSE }, "
141 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
145 gst_alsasrc_finalize (GObject * object)
147 GstAlsaSrc *src = GST_ALSA_SRC (object);
149 g_free (src->device);
150 g_mutex_free (src->alsa_lock);
152 G_OBJECT_CLASS (parent_class)->finalize (object);
156 gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type)
158 /* only support this one interface (wrapped by GstImplementsInterface) */
159 g_assert (interface_type == GST_TYPE_MIXER);
161 return gst_alsasrc_mixer_supported (this, interface_type);
165 gst_implements_interface_init (GstImplementsInterfaceClass * klass)
167 klass->supported = (gpointer) gst_alsasrc_interface_supported;
171 gst_alsasrc_init_interfaces (GType type)
173 static const GInterfaceInfo implements_iface_info = {
174 (GInterfaceInitFunc) gst_implements_interface_init,
178 static const GInterfaceInfo mixer_iface_info = {
179 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,
184 g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
185 &implements_iface_info);
186 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
188 gst_alsa_type_add_device_property_probe_interface (type);
192 gst_alsasrc_base_init (gpointer g_class)
194 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
196 gst_element_class_set_details_simple (element_class,
197 "Audio source (ALSA)", "Source/Audio",
198 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
200 gst_element_class_add_static_pad_template (element_class,
201 &alsasrc_src_factory);
205 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
207 GObjectClass *gobject_class;
208 GstElementClass *gstelement_class;
209 GstBaseSrcClass *gstbasesrc_class;
210 GstAudioSrcClass *gstaudiosrc_class;
212 gobject_class = (GObjectClass *) klass;
213 gstelement_class = (GstElementClass *) klass;
214 gstbasesrc_class = (GstBaseSrcClass *) klass;
215 gstaudiosrc_class = (GstAudioSrcClass *) klass;
217 gobject_class->finalize = gst_alsasrc_finalize;
218 gobject_class->get_property = gst_alsasrc_get_property;
219 gobject_class->set_property = gst_alsasrc_set_property;
221 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
223 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
224 gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_alsasrc_create);
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);
234 g_object_class_install_property (gobject_class, PROP_DEVICE,
235 g_param_spec_string ("device", "Device",
236 "ALSA device, as defined in an asound configuration file",
237 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
239 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
240 g_param_spec_string ("device-name", "Device name",
241 "Human-readable name of the sound device",
242 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
244 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
245 g_param_spec_string ("card-name", "Card name",
246 "Human-readable name of the sound card",
247 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
251 gst_alsasrc_get_timestamp (GstAlsaSrc * src)
253 snd_pcm_status_t *status;
254 snd_htimestamp_t tstamp;
255 GstClockTime timestamp;
256 snd_pcm_uframes_t availmax;
259 GST_DEBUG_OBJECT (src, "Getting alsa timestamp!");
262 GST_ERROR_OBJECT (src, "No alsa handle created yet !");
263 return GST_CLOCK_TIME_NONE;
266 if (snd_pcm_status_malloc (&status) != 0) {
267 GST_ERROR_OBJECT (src, "snd_pcm_status_malloc failed");
268 return GST_CLOCK_TIME_NONE;
271 if (snd_pcm_status (src->handle, status) != 0) {
272 GST_ERROR_OBJECT (src, "snd_pcm_status failed");
273 snd_pcm_status_free (status);
274 return GST_CLOCK_TIME_NONE;
277 /* get high resolution time stamp from driver */
278 snd_pcm_status_get_htstamp (status, &tstamp);
279 timestamp = GST_TIMESPEC_TO_TIME (tstamp);
280 GST_DEBUG_OBJECT (src, "Base ts: %" GST_TIME_FORMAT,
281 GST_TIME_ARGS (timestamp));
282 if (timestamp == 0) {
283 /* This timestamp is supposed to represent the last sample, so 0 (which
284 can be returned on some ALSA setups (such as mine)) must mean that it
285 is invalid, unless there's just one sample, but we'll ignore that. */
286 GST_WARNING_OBJECT (src,
287 "No timestamp returned from snd_pcm_status_get_htstamp");
288 return GST_CLOCK_TIME_NONE;
291 /* Max available frames sets the depth of the buffer */
292 availmax = snd_pcm_status_get_avail_max (status);
294 /* Compensate the fact that the timestamp references the last sample */
295 offset = -gst_util_uint64_scale_int (availmax * 2, GST_SECOND, src->rate);
296 /* Compensate for the delay until the package is available */
297 offset += gst_util_uint64_scale_int (snd_pcm_status_get_delay (status),
298 GST_SECOND, src->rate);
300 snd_pcm_status_free (status);
302 /* just in case, should not happen */
303 if (-offset > timestamp)
308 /* Take first ts into account */
309 if (src->first_alsa_ts == GST_CLOCK_TIME_NONE) {
310 src->first_alsa_ts = timestamp;
312 timestamp -= src->first_alsa_ts;
314 GST_DEBUG_OBJECT (src, "ALSA timestamp : %" GST_TIME_FORMAT,
315 GST_TIME_ARGS (timestamp));
320 gst_alsasrc_set_property (GObject * object, guint prop_id,
321 const GValue * value, GParamSpec * pspec)
325 src = GST_ALSA_SRC (object);
329 g_free (src->device);
330 src->device = g_value_dup_string (value);
331 if (src->device == NULL) {
332 src->device = g_strdup (DEFAULT_PROP_DEVICE);
336 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
342 gst_alsasrc_get_property (GObject * object, guint prop_id,
343 GValue * value, GParamSpec * pspec)
347 src = GST_ALSA_SRC (object);
351 g_value_set_string (value, src->device);
353 case PROP_DEVICE_NAME:
354 g_value_take_string (value,
355 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
356 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
359 g_value_take_string (value,
360 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
361 src->device, SND_PCM_STREAM_CAPTURE));
364 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
369 static GstStateChangeReturn
370 gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
372 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
373 GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
374 GstAlsaSrc *asrc = GST_ALSA_SRC (element);
377 switch (transition) {
378 /* Show the compiler that we care */
379 case GST_STATE_CHANGE_NULL_TO_READY:
380 case GST_STATE_CHANGE_READY_TO_PAUSED:
381 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
382 case GST_STATE_CHANGE_PAUSED_TO_READY:
383 case GST_STATE_CHANGE_READY_TO_NULL:
386 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
388 asrc->driver_timestamps = FALSE;
389 if (GST_IS_SYSTEM_CLOCK (clk)) {
391 g_object_get (clk, "clock-type", &clocktype, NULL);
392 if (clocktype == GST_CLOCK_TYPE_MONOTONIC) {
393 asrc->driver_timestamps = TRUE;
398 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
404 gst_alsasrc_create (GstBaseSrc * bsrc, guint64 offset, guint length,
407 GstFlowReturn ret = GST_FLOW_OK;
408 GstAlsaSrc *asrc = GST_ALSA_SRC (bsrc);
411 GST_BASE_SRC_CLASS (parent_class)->create (bsrc, offset, length, outbuf);
412 if (asrc->driver_timestamps == TRUE && *outbuf) {
413 GstClockTime ts = gst_alsasrc_get_timestamp (asrc);
414 if (GST_CLOCK_TIME_IS_VALID (ts)) {
415 GST_BUFFER_TIMESTAMP (*outbuf) = ts;
423 gst_alsasrc_init (GstAlsaSrc * alsasrc, GstAlsaSrcClass * g_class)
425 GST_DEBUG_OBJECT (alsasrc, "initializing");
427 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
428 alsasrc->cached_caps = NULL;
429 alsasrc->driver_timestamps = FALSE;
430 alsasrc->first_alsa_ts = GST_CLOCK_TIME_NONE;
432 alsasrc->alsa_lock = g_mutex_new ();
435 #define CHECK(call, error) \
437 if ((err = call) < 0) \
443 gst_alsasrc_getcaps (GstBaseSrc * bsrc)
445 GstElementClass *element_class;
446 GstPadTemplate *pad_template;
450 src = GST_ALSA_SRC (bsrc);
452 if (src->handle == NULL) {
453 GST_DEBUG_OBJECT (src, "device not open, using template caps");
454 return NULL; /* base class will get template caps for us */
457 if (src->cached_caps) {
458 GST_LOG_OBJECT (src, "Returning cached caps");
459 return gst_caps_ref (src->cached_caps);
462 element_class = GST_ELEMENT_GET_CLASS (src);
463 pad_template = gst_element_class_get_pad_template (element_class, "src");
464 g_return_val_if_fail (pad_template != NULL, NULL);
466 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle,
467 gst_pad_template_get_caps (pad_template));
470 src->cached_caps = gst_caps_ref (caps);
473 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
479 set_hwparams (GstAlsaSrc * alsa)
483 snd_pcm_hw_params_t *params;
485 snd_pcm_hw_params_malloc (¶ms);
487 /* choose all parameters */
488 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
489 /* set the interleaved read/write format */
490 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
492 /* set the sample format */
493 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
495 /* set the count of channels */
496 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
498 /* set the stream rate */
500 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
502 if (rrate != alsa->rate)
505 if (alsa->buffer_time != -1) {
506 /* set the buffer time */
507 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
508 &alsa->buffer_time, NULL), buffer_time);
510 if (alsa->period_time != -1) {
511 /* set the period time */
512 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
513 &alsa->period_time, NULL), period_time);
516 /* write the parameters to device */
517 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
519 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
522 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
525 snd_pcm_hw_params_free (params);
531 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
532 ("Broken configuration for recording: no configurations available: %s",
533 snd_strerror (err)));
534 snd_pcm_hw_params_free (params);
539 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
540 ("Access type not available for recording: %s", snd_strerror (err)));
541 snd_pcm_hw_params_free (params);
546 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
547 ("Sample format not available for recording: %s", snd_strerror (err)));
548 snd_pcm_hw_params_free (params);
555 if ((alsa->channels) == 1)
556 msg = g_strdup (_("Could not open device for recording in mono mode."));
557 if ((alsa->channels) == 2)
558 msg = g_strdup (_("Could not open device for recording in stereo mode."));
559 if ((alsa->channels) > 2)
562 ("Could not open device for recording in %d-channel mode"),
564 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
565 ("%s", snd_strerror (err)));
567 snd_pcm_hw_params_free (params);
572 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
573 ("Rate %iHz not available for recording: %s",
574 alsa->rate, snd_strerror (err)));
575 snd_pcm_hw_params_free (params);
580 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
581 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
582 snd_pcm_hw_params_free (params);
587 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
588 ("Unable to set buffer time %i for recording: %s",
589 alsa->buffer_time, snd_strerror (err)));
590 snd_pcm_hw_params_free (params);
595 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
596 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
597 snd_pcm_hw_params_free (params);
602 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
603 ("Unable to set period time %i for recording: %s", alsa->period_time,
604 snd_strerror (err)));
605 snd_pcm_hw_params_free (params);
610 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
611 ("Unable to get period size for recording: %s", snd_strerror (err)));
612 snd_pcm_hw_params_free (params);
617 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
618 ("Unable to set hw params for recording: %s", snd_strerror (err)));
619 snd_pcm_hw_params_free (params);
625 set_swparams (GstAlsaSrc * alsa)
628 snd_pcm_sw_params_t *params;
630 snd_pcm_sw_params_malloc (¶ms);
632 /* get the current swparams */
633 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
634 /* allow the transfer when at least period_size samples can be processed */
635 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
636 alsa->period_size), set_avail);
637 /* start the transfer on first read */
638 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
639 0), start_threshold);
641 #if GST_CHECK_ALSA_VERSION(1,0,16)
642 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
644 /* align all transfers to 1 sample */
645 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
648 /* write the parameters to the recording device */
649 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
651 snd_pcm_sw_params_free (params);
657 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
658 ("Unable to determine current swparams for playback: %s",
659 snd_strerror (err)));
660 snd_pcm_sw_params_free (params);
665 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
666 ("Unable to set start threshold mode for playback: %s",
667 snd_strerror (err)));
668 snd_pcm_sw_params_free (params);
673 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
674 ("Unable to set avail min for playback: %s", snd_strerror (err)));
675 snd_pcm_sw_params_free (params);
678 #if !GST_CHECK_ALSA_VERSION(1,0,16)
681 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
682 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
683 snd_pcm_sw_params_free (params);
689 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
690 ("Unable to set sw params for playback: %s", snd_strerror (err)));
691 snd_pcm_sw_params_free (params);
697 alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec)
699 switch (spec->type) {
700 case GST_BUFTYPE_LINEAR:
701 alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
702 spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
704 case GST_BUFTYPE_FLOAT:
705 switch (spec->format) {
707 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
710 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
713 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
716 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
722 case GST_BUFTYPE_A_LAW:
723 alsa->format = SND_PCM_FORMAT_A_LAW;
725 case GST_BUFTYPE_MU_LAW:
726 alsa->format = SND_PCM_FORMAT_MU_LAW;
732 alsa->rate = spec->rate;
733 alsa->channels = spec->channels;
734 alsa->buffer_time = spec->buffer_time;
735 alsa->period_time = spec->latency_time;
736 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
748 gst_alsasrc_open (GstAudioSrc * asrc)
753 alsa = GST_ALSA_SRC (asrc);
755 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
756 SND_PCM_NONBLOCK), open_error);
759 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE);
767 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
768 (_("Could not open audio device for recording. "
769 "Device is being used by another application.")),
770 ("Device '%s' is busy", alsa->device));
772 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
773 (_("Could not open audio device for recording.")),
774 ("Recording open error on device '%s': %s", alsa->device,
775 snd_strerror (err)));
782 gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
787 alsa = GST_ALSA_SRC (asrc);
789 if (!alsasrc_parse_spec (alsa, spec))
792 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
794 CHECK (set_hwparams (alsa), hw_params_failed);
795 CHECK (set_swparams (alsa), sw_params_failed);
796 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
798 alsa->bytes_per_sample = spec->bytes_per_sample;
799 spec->segsize = alsa->period_size * spec->bytes_per_sample;
800 spec->segtotal = alsa->buffer_size / alsa->period_size;
801 spec->silence_sample[0] = 0;
802 spec->silence_sample[1] = 0;
803 spec->silence_sample[2] = 0;
804 spec->silence_sample[3] = 0;
811 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
812 ("Error parsing spec"));
817 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
818 ("Could not set device to blocking: %s", snd_strerror (err)));
823 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
824 ("Setting of hwparams failed: %s", snd_strerror (err)));
829 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
830 ("Setting of swparams failed: %s", snd_strerror (err)));
835 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
836 ("Prepare failed: %s", snd_strerror (err)));
842 gst_alsasrc_unprepare (GstAudioSrc * asrc)
846 alsa = GST_ALSA_SRC (asrc);
848 snd_pcm_drop (alsa->handle);
849 snd_pcm_hw_free (alsa->handle);
850 snd_pcm_nonblock (alsa->handle, 1);
856 gst_alsasrc_close (GstAudioSrc * asrc)
858 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
860 snd_pcm_close (alsa->handle);
864 gst_alsa_mixer_free (alsa->mixer);
868 gst_caps_replace (&alsa->cached_caps, NULL);
874 * Underrun and suspend recovery
877 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
879 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
881 if (err == -EPIPE) { /* under-run */
882 err = snd_pcm_prepare (handle);
884 GST_WARNING_OBJECT (alsa,
885 "Can't recovery from underrun, prepare failed: %s",
888 } else if (err == -ESTRPIPE) {
889 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
890 g_usleep (100); /* wait until the suspend flag is released */
893 err = snd_pcm_prepare (handle);
895 GST_WARNING_OBJECT (alsa,
896 "Can't recovery from suspend, prepare failed: %s",
905 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length)
912 alsa = GST_ALSA_SRC (asrc);
914 cptr = length / alsa->bytes_per_sample;
917 GST_ALSA_SRC_LOCK (asrc);
919 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
920 if (err == -EAGAIN) {
921 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
923 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
929 ptr += err * alsa->channels;
932 GST_ALSA_SRC_UNLOCK (asrc);
934 return length - (cptr * alsa->bytes_per_sample);
938 GST_ALSA_SRC_UNLOCK (asrc);
939 return length; /* skip one period */
944 gst_alsasrc_delay (GstAudioSrc * asrc)
947 snd_pcm_sframes_t delay;
950 alsa = GST_ALSA_SRC (asrc);
952 res = snd_pcm_delay (alsa->handle, &delay);
953 if (G_UNLIKELY (res < 0)) {
954 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
958 return CLAMP (delay, 0, alsa->buffer_size);
962 gst_alsasrc_reset (GstAudioSrc * asrc)
967 alsa = GST_ALSA_SRC (asrc);
969 GST_ALSA_SRC_LOCK (asrc);
970 GST_DEBUG_OBJECT (alsa, "drop");
971 CHECK (snd_pcm_drop (alsa->handle), drop_error);
972 GST_DEBUG_OBJECT (alsa, "prepare");
973 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
974 GST_DEBUG_OBJECT (alsa, "reset done");
975 alsa->first_alsa_ts = GST_CLOCK_TIME_NONE;
976 GST_ALSA_SRC_UNLOCK (asrc);
983 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
985 GST_ALSA_SRC_UNLOCK (asrc);
990 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
992 GST_ALSA_SRC_UNLOCK (asrc);