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., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * SECTION:element-alsasrc
27 * This element reads data from an audio card using the ALSA API.
29 * ## Example pipelines
31 * gst-launch-1.0 -v alsasrc ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
33 * Record from a sound card using ALSA and encode to Ogg/Vorbis.
40 #include <sys/ioctl.h>
46 #include <alsa/asoundlib.h>
48 #include "gstalsaelements.h"
49 #include "gstalsasrc.h"
51 #include <gst/gst-i18n-plugin.h>
54 #define ESTRPIPE EPIPE
57 #define DEFAULT_PROP_DEVICE "default"
58 #define DEFAULT_PROP_DEVICE_NAME ""
59 #define DEFAULT_PROP_CARD_NAME ""
60 #define DEFAULT_PROP_USE_DRIVER_TIMESTAMP TRUE
68 PROP_USE_DRIVER_TIMESTAMP,
72 #define gst_alsasrc_parent_class parent_class
73 G_DEFINE_TYPE (GstAlsaSrc, gst_alsasrc, GST_TYPE_AUDIO_SRC);
74 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (alsasrc, "alsasrc", GST_RANK_PRIMARY,
75 GST_TYPE_ALSA_SRC, alsa_element_init (plugin));
77 static void gst_alsasrc_finalize (GObject * object);
78 static void gst_alsasrc_set_property (GObject * object,
79 guint prop_id, const GValue * value, GParamSpec * pspec);
80 static void gst_alsasrc_get_property (GObject * object,
81 guint prop_id, GValue * value, GParamSpec * pspec);
82 static GstStateChangeReturn gst_alsasrc_change_state (GstElement * element,
83 GstStateChange transition);
84 static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter);
86 static gboolean gst_alsasrc_open (GstAudioSrc * asrc);
87 static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,
88 GstAudioRingBufferSpec * spec);
89 static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);
90 static gboolean gst_alsasrc_close (GstAudioSrc * asrc);
91 static guint gst_alsasrc_read
92 (GstAudioSrc * asrc, gpointer data, guint length, GstClockTime * timestamp);
93 static guint gst_alsasrc_delay (GstAudioSrc * asrc);
94 static void gst_alsasrc_reset (GstAudioSrc * asrc);
96 /* AlsaSrc signals and args */
102 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
103 # define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
105 # define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
108 static GstStaticPadTemplate alsasrc_src_factory =
109 GST_STATIC_PAD_TEMPLATE ("src",
112 GST_STATIC_CAPS ("audio/x-raw, "
113 "format = (string) " GST_AUDIO_FORMATS_ALL ", "
114 "layout = (string) interleaved, "
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_clear (&src->alsa_lock);
126 G_OBJECT_CLASS (parent_class)->finalize (object);
130 gst_alsasrc_class_init (GstAlsaSrcClass * klass)
132 GObjectClass *gobject_class;
133 GstElementClass *gstelement_class;
134 GstBaseSrcClass *gstbasesrc_class;
135 GstAudioSrcClass *gstaudiosrc_class;
137 gobject_class = (GObjectClass *) klass;
138 gstelement_class = (GstElementClass *) klass;
139 gstbasesrc_class = (GstBaseSrcClass *) klass;
140 gstaudiosrc_class = (GstAudioSrcClass *) klass;
142 gobject_class->finalize = gst_alsasrc_finalize;
143 gobject_class->get_property = gst_alsasrc_get_property;
144 gobject_class->set_property = gst_alsasrc_set_property;
146 gst_element_class_set_static_metadata (gstelement_class,
147 "Audio source (ALSA)", "Source/Audio",
148 "Read from a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
150 gst_element_class_add_static_pad_template (gstelement_class,
151 &alsasrc_src_factory);
153 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);
155 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);
156 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);
157 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);
158 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);
159 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);
160 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);
161 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset);
162 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_alsasrc_change_state);
164 g_object_class_install_property (gobject_class, PROP_DEVICE,
165 g_param_spec_string ("device", "Device",
166 "ALSA device, as defined in an asound configuration file",
167 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
169 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
170 g_param_spec_string ("device-name", "Device name",
171 "Human-readable name of the sound device",
172 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
174 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
175 g_param_spec_string ("card-name", "Card name",
176 "Human-readable name of the sound card",
177 DEFAULT_PROP_CARD_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS
178 | GST_PARAM_DOC_SHOW_DEFAULT));
180 g_object_class_install_property (gobject_class, PROP_USE_DRIVER_TIMESTAMP,
181 g_param_spec_boolean ("use-driver-timestamps", "Use driver timestamps",
182 "Use driver timestamps or the pipeline clock timestamps",
183 DEFAULT_PROP_USE_DRIVER_TIMESTAMP,
184 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
188 gst_alsasrc_set_property (GObject * object, guint prop_id,
189 const GValue * value, GParamSpec * pspec)
193 src = GST_ALSA_SRC (object);
197 g_free (src->device);
198 src->device = g_value_dup_string (value);
199 if (src->device == NULL) {
200 src->device = g_strdup (DEFAULT_PROP_DEVICE);
203 case PROP_USE_DRIVER_TIMESTAMP:
204 GST_OBJECT_LOCK (src);
205 src->use_driver_timestamps = g_value_get_boolean (value);
206 GST_OBJECT_UNLOCK (src);
209 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
215 gst_alsasrc_get_property (GObject * object, guint prop_id,
216 GValue * value, GParamSpec * pspec)
220 src = GST_ALSA_SRC (object);
224 g_value_set_string (value, src->device);
226 case PROP_DEVICE_NAME:
227 g_value_take_string (value,
228 gst_alsa_find_device_name (GST_OBJECT_CAST (src),
229 src->device, src->handle, SND_PCM_STREAM_CAPTURE));
232 g_value_take_string (value,
233 gst_alsa_find_card_name (GST_OBJECT_CAST (src),
234 src->device, SND_PCM_STREAM_CAPTURE));
236 case PROP_USE_DRIVER_TIMESTAMP:
237 GST_OBJECT_LOCK (src);
238 g_value_set_boolean (value, src->use_driver_timestamps);
239 GST_OBJECT_UNLOCK (src);
242 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
247 static GstStateChangeReturn
248 gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
250 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
251 GstAlsaSrc *alsa = GST_ALSA_SRC (element);
254 switch (transition) {
255 /* show the compiler that we care */
256 case GST_STATE_CHANGE_NULL_TO_READY:
257 case GST_STATE_CHANGE_READY_TO_PAUSED:
258 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
259 case GST_STATE_CHANGE_PAUSED_TO_READY:
260 case GST_STATE_CHANGE_READY_TO_NULL:
261 case GST_STATE_CHANGE_NULL_TO_NULL:
262 case GST_STATE_CHANGE_READY_TO_READY:
263 case GST_STATE_CHANGE_PAUSED_TO_PAUSED:
264 case GST_STATE_CHANGE_PLAYING_TO_PLAYING:
266 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
267 alsa->driver_timestamps = FALSE;
269 clk = gst_element_get_clock (element);
271 if (G_OBJECT_TYPE (clk) == GST_TYPE_SYSTEM_CLOCK) {
273 g_object_get (clk, "clock-type", &clocktype, NULL);
274 if (clocktype == GST_CLOCK_TYPE_MONOTONIC &&
275 alsa->use_driver_timestamps) {
276 GST_INFO ("Using driver timestamps !");
277 alsa->driver_timestamps = TRUE;
279 GST_INFO ("Not using driver timestamps !");
280 alsa->driver_timestamps = FALSE;
284 gst_object_unref (clk);
288 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
294 gst_alsasrc_init (GstAlsaSrc * alsasrc)
296 GST_DEBUG_OBJECT (alsasrc, "initializing");
298 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);
299 alsasrc->cached_caps = NULL;
300 alsasrc->driver_timestamps = FALSE;
301 alsasrc->use_driver_timestamps = DEFAULT_PROP_USE_DRIVER_TIMESTAMP;
303 g_mutex_init (&alsasrc->alsa_lock);
306 #define CHECK(call, error) \
308 if ((err = call) < 0) \
314 gst_alsasrc_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
316 GstElementClass *element_class;
317 GstPadTemplate *pad_template;
319 GstCaps *caps, *templ_caps;
321 src = GST_ALSA_SRC (bsrc);
323 if (src->handle == NULL) {
324 GST_DEBUG_OBJECT (src, "device not open, using template caps");
325 return GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
328 if (src->cached_caps) {
329 GST_LOG_OBJECT (src, "Returning cached caps");
331 return gst_caps_intersect_full (filter, src->cached_caps,
332 GST_CAPS_INTERSECT_FIRST);
334 return gst_caps_ref (src->cached_caps);
337 element_class = GST_ELEMENT_GET_CLASS (src);
338 pad_template = gst_element_class_get_pad_template (element_class, "src");
339 g_return_val_if_fail (pad_template != NULL, NULL);
341 templ_caps = gst_pad_template_get_caps (pad_template);
342 GST_INFO_OBJECT (src, "template caps %" GST_PTR_FORMAT, templ_caps);
344 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src),
345 src->device, src->handle, templ_caps);
346 gst_caps_unref (templ_caps);
349 src->cached_caps = gst_caps_ref (caps);
352 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps);
355 GstCaps *intersection;
358 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
359 gst_caps_unref (caps);
367 set_hwparams (GstAlsaSrc * alsa)
371 snd_pcm_hw_params_t *params, *params_copy;
373 snd_pcm_hw_params_malloc (¶ms);
374 snd_pcm_hw_params_malloc (¶ms_copy);
376 /* choose all parameters */
377 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
378 /* set the interleaved read/write format */
379 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
381 /* set the sample format */
382 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
384 /* set the count of channels */
385 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
387 /* set the stream rate */
389 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
391 if (rrate != alsa->rate)
394 #ifndef GST_DISABLE_GST_DEBUG
395 /* get and dump some limits */
399 snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL);
400 snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);
402 GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",
403 alsa->buffer_time, min, max);
405 snd_pcm_hw_params_get_period_time_min (params, &min, NULL);
406 snd_pcm_hw_params_get_period_time_max (params, &max, NULL);
408 GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",
409 alsa->period_time, min, max);
411 snd_pcm_hw_params_get_periods_min (params, &min, NULL);
412 snd_pcm_hw_params_get_periods_max (params, &max, NULL);
414 GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
417 /* Keep a copy of initial params struct that can be used later */
418 snd_pcm_hw_params_copy (params_copy, params);
419 /* Following pulseaudio's approach in
420 * https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/557c4295107dc7374c850b0bd5331dd35e8fdd0f
421 * we'll try various configuration to set the buffer time and period time as some
422 * driver can be picky on the order of the calls.
424 if (alsa->period_time != -1 && alsa->buffer_time != -1) {
425 if ((snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
426 &alsa->period_time, NULL) >= 0)
427 && (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
428 &alsa->buffer_time, NULL) >= 0)) {
429 GST_DEBUG_OBJECT (alsa, "period time %u buffer time %u set correctly",
430 alsa->period_time, alsa->buffer_time);
433 /* Try the new order with previous params struct as current one might
434 have partial settings from the order that was tried unsuccessfully */
435 snd_pcm_hw_params_copy (params, params_copy);
436 if ((snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
437 &alsa->buffer_time, NULL) >= 0)
438 && (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
439 &alsa->period_time, NULL) >= 0)) {
440 GST_DEBUG_OBJECT (alsa, "buffer time %u period time %u set correctly",
441 alsa->buffer_time, alsa->period_time);
445 if (alsa->period_time != -1) {
446 snd_pcm_hw_params_copy (params, params_copy);
447 /* set the period time only */
448 if ((snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
449 &alsa->period_time, NULL) >= 0)) {
450 GST_DEBUG_OBJECT (alsa, "period time %u set correctly",
455 if (alsa->buffer_time != -1) {
456 snd_pcm_hw_params_copy (params, params_copy);
457 /* set the buffer time only */
458 if ((snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
459 &alsa->buffer_time, NULL) >= 0)) {
460 GST_DEBUG_OBJECT (alsa, "buffer time %u set correctly",
465 /* Set nothing if all above failed */
466 snd_pcm_hw_params_copy (params, params_copy);
467 GST_DEBUG_OBJECT (alsa, "Not setting period time and buffer time");
470 /* write the parameters to device */
471 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
472 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
474 GST_DEBUG_OBJECT (alsa, "buffer size : %lu", alsa->buffer_size);
475 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size,
477 GST_DEBUG_OBJECT (alsa, "period size : %lu", alsa->period_size);
483 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
484 ("Broken configuration for recording: no configurations available: %s",
485 snd_strerror (err)));
490 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
491 ("Access type not available for recording: %s", snd_strerror (err)));
496 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
497 ("Sample format not available for recording: %s", snd_strerror (err)));
504 if ((alsa->channels) == 1)
505 msg = g_strdup (_("Could not open device for recording in mono mode."));
506 if ((alsa->channels) == 2)
507 msg = g_strdup (_("Could not open device for recording in stereo mode."));
508 if ((alsa->channels) > 2)
511 ("Could not open device for recording in %d-channel mode"),
513 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
514 ("%s", snd_strerror (err)));
520 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
521 ("Rate %iHz not available for recording: %s",
522 alsa->rate, snd_strerror (err)));
527 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
528 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
534 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
535 ("Unable to get buffer size for recording: %s", snd_strerror (err)));
540 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
541 ("Unable to get period size for recording: %s", snd_strerror (err)));
546 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
547 ("Unable to set hw params for recording: %s", snd_strerror (err)));
551 snd_pcm_hw_params_free (params);
552 snd_pcm_hw_params_free (params_copy);
558 set_swparams (GstAlsaSrc * alsa)
561 snd_pcm_sw_params_t *params;
563 snd_pcm_sw_params_malloc (¶ms);
565 /* get the current swparams */
566 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
567 /* allow the transfer when at least period_size samples can be processed */
568 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
569 alsa->period_size), set_avail);
570 /* start the transfer on first read */
571 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
572 0), start_threshold);
573 /* use monotonic timestamping */
574 CHECK (snd_pcm_sw_params_set_tstamp_mode (alsa->handle, params,
575 SND_PCM_TSTAMP_MMAP), tstamp_mode);
577 #if GST_CHECK_ALSA_VERSION(1,0,16)
578 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
580 /* align all transfers to 1 sample */
581 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
584 /* write the parameters to the recording device */
585 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
587 snd_pcm_sw_params_free (params);
593 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
594 ("Unable to determine current swparams for playback: %s",
595 snd_strerror (err)));
596 snd_pcm_sw_params_free (params);
601 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
602 ("Unable to set start threshold mode for playback: %s",
603 snd_strerror (err)));
604 snd_pcm_sw_params_free (params);
609 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
610 ("Unable to set avail min for playback: %s", snd_strerror (err)));
611 snd_pcm_sw_params_free (params);
616 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
617 ("Unable to set tstamp mode for playback: %s", snd_strerror (err)));
618 snd_pcm_sw_params_free (params);
621 #if !GST_CHECK_ALSA_VERSION(1,0,16)
624 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
625 ("Unable to set transfer align for playback: %s", snd_strerror (err)));
626 snd_pcm_sw_params_free (params);
632 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
633 ("Unable to set sw params for playback: %s", snd_strerror (err)));
634 snd_pcm_sw_params_free (params);
640 alsasrc_parse_spec (GstAlsaSrc * alsa, GstAudioRingBufferSpec * spec)
642 switch (spec->type) {
643 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
644 switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
645 case GST_AUDIO_FORMAT_U8:
646 alsa->format = SND_PCM_FORMAT_U8;
648 case GST_AUDIO_FORMAT_S8:
649 alsa->format = SND_PCM_FORMAT_S8;
651 case GST_AUDIO_FORMAT_S16LE:
652 alsa->format = SND_PCM_FORMAT_S16_LE;
654 case GST_AUDIO_FORMAT_S16BE:
655 alsa->format = SND_PCM_FORMAT_S16_BE;
657 case GST_AUDIO_FORMAT_U16LE:
658 alsa->format = SND_PCM_FORMAT_U16_LE;
660 case GST_AUDIO_FORMAT_U16BE:
661 alsa->format = SND_PCM_FORMAT_U16_BE;
663 case GST_AUDIO_FORMAT_S24_32LE:
664 alsa->format = SND_PCM_FORMAT_S24_LE;
666 case GST_AUDIO_FORMAT_S24_32BE:
667 alsa->format = SND_PCM_FORMAT_S24_BE;
669 case GST_AUDIO_FORMAT_U24_32LE:
670 alsa->format = SND_PCM_FORMAT_U24_LE;
672 case GST_AUDIO_FORMAT_U24_32BE:
673 alsa->format = SND_PCM_FORMAT_U24_BE;
675 case GST_AUDIO_FORMAT_S32LE:
676 alsa->format = SND_PCM_FORMAT_S32_LE;
678 case GST_AUDIO_FORMAT_S32BE:
679 alsa->format = SND_PCM_FORMAT_S32_BE;
681 case GST_AUDIO_FORMAT_U32LE:
682 alsa->format = SND_PCM_FORMAT_U32_LE;
684 case GST_AUDIO_FORMAT_U32BE:
685 alsa->format = SND_PCM_FORMAT_U32_BE;
687 case GST_AUDIO_FORMAT_S24LE:
688 alsa->format = SND_PCM_FORMAT_S24_3LE;
690 case GST_AUDIO_FORMAT_S24BE:
691 alsa->format = SND_PCM_FORMAT_S24_3BE;
693 case GST_AUDIO_FORMAT_U24LE:
694 alsa->format = SND_PCM_FORMAT_U24_3LE;
696 case GST_AUDIO_FORMAT_U24BE:
697 alsa->format = SND_PCM_FORMAT_U24_3BE;
699 case GST_AUDIO_FORMAT_S20LE:
700 alsa->format = SND_PCM_FORMAT_S20_3LE;
702 case GST_AUDIO_FORMAT_S20BE:
703 alsa->format = SND_PCM_FORMAT_S20_3BE;
705 case GST_AUDIO_FORMAT_U20LE:
706 alsa->format = SND_PCM_FORMAT_U20_3LE;
708 case GST_AUDIO_FORMAT_U20BE:
709 alsa->format = SND_PCM_FORMAT_U20_3BE;
711 case GST_AUDIO_FORMAT_S18LE:
712 alsa->format = SND_PCM_FORMAT_S18_3LE;
714 case GST_AUDIO_FORMAT_S18BE:
715 alsa->format = SND_PCM_FORMAT_S18_3BE;
717 case GST_AUDIO_FORMAT_U18LE:
718 alsa->format = SND_PCM_FORMAT_U18_3LE;
720 case GST_AUDIO_FORMAT_U18BE:
721 alsa->format = SND_PCM_FORMAT_U18_3BE;
723 case GST_AUDIO_FORMAT_F32LE:
724 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
726 case GST_AUDIO_FORMAT_F32BE:
727 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
729 case GST_AUDIO_FORMAT_F64LE:
730 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
732 case GST_AUDIO_FORMAT_F64BE:
733 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
739 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
740 alsa->format = SND_PCM_FORMAT_A_LAW;
742 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
743 alsa->format = SND_PCM_FORMAT_MU_LAW;
749 alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
750 alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
751 alsa->buffer_time = spec->buffer_time;
752 alsa->period_time = spec->latency_time;
753 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
755 if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9)
756 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
757 (alsa)->ringbuffer, alsa_position[alsa->channels - 1]);
769 gst_alsasrc_open (GstAudioSrc * asrc)
774 alsa = GST_ALSA_SRC (asrc);
776 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE,
777 (alsa->driver_timestamps) ? 0 : SND_PCM_NONBLOCK), open_error);
785 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
786 (_("Could not open audio device for recording. "
787 "Device is being used by another application.")),
788 ("Device '%s' is busy", alsa->device));
790 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ,
791 (_("Could not open audio device for recording.")),
792 ("Recording open error on device '%s': %s", alsa->device,
793 snd_strerror (err)));
800 gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
805 alsa = GST_ALSA_SRC (asrc);
807 if (!alsasrc_parse_spec (alsa, spec))
810 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
812 CHECK (set_hwparams (alsa), hw_params_failed);
813 CHECK (set_swparams (alsa), sw_params_failed);
814 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
816 alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
817 spec->segsize = alsa->period_size * alsa->bpf;
818 spec->segtotal = alsa->buffer_size / alsa->period_size;
821 snd_output_t *out_buf = NULL;
824 snd_output_buffer_open (&out_buf);
825 snd_pcm_dump_hw_setup (alsa->handle, out_buf);
826 snd_output_buffer_string (out_buf, &msg);
827 GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
828 snd_output_close (out_buf);
829 snd_output_buffer_open (&out_buf);
830 snd_pcm_dump_sw_setup (alsa->handle, out_buf);
831 snd_output_buffer_string (out_buf, &msg);
832 GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
833 snd_output_close (out_buf);
836 #ifdef SND_CHMAP_API_VERSION
837 alsa_detect_channels_mapping (GST_OBJECT (alsa), alsa->handle, spec,
838 alsa->channels, GST_AUDIO_BASE_SRC (alsa)->ringbuffer);
839 #endif /* SND_CHMAP_API_VERSION */
846 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
847 ("Error parsing spec"));
852 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
853 ("Could not set device to blocking: %s", snd_strerror (err)));
858 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
859 ("Setting of hwparams failed: %s", snd_strerror (err)));
864 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
865 ("Setting of swparams failed: %s", snd_strerror (err)));
870 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
871 ("Prepare failed: %s", snd_strerror (err)));
877 gst_alsasrc_unprepare (GstAudioSrc * asrc)
881 alsa = GST_ALSA_SRC (asrc);
883 snd_pcm_drop (alsa->handle);
884 snd_pcm_hw_free (alsa->handle);
885 snd_pcm_nonblock (alsa->handle, 1);
891 gst_alsasrc_close (GstAudioSrc * asrc)
893 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc);
895 snd_pcm_close (alsa->handle);
898 gst_caps_replace (&alsa->cached_caps, NULL);
904 * Underrun and suspend recovery
907 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err)
909 GST_WARNING_OBJECT (alsa, "xrun recovery %d: %s", err, g_strerror (-err));
911 if (err == -EPIPE) { /* under-run */
912 err = snd_pcm_prepare (handle);
914 GST_WARNING_OBJECT (alsa,
915 "Can't recover from underrun, prepare failed: %s",
918 } else if (err == -ESTRPIPE) {
919 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
920 g_usleep (100); /* wait until the suspend flag is released */
923 err = snd_pcm_prepare (handle);
925 GST_WARNING_OBJECT (alsa,
926 "Can't recover from suspend, prepare failed: %s",
935 gst_alsasrc_get_timestamp (GstAlsaSrc * asrc)
937 snd_pcm_status_t *status;
938 snd_htimestamp_t tstamp;
939 GstClockTime timestamp;
940 snd_pcm_uframes_t avail;
943 if (G_UNLIKELY (!asrc)) {
944 GST_ERROR_OBJECT (asrc, "No alsa handle created yet !");
945 return GST_CLOCK_TIME_NONE;
948 if (G_UNLIKELY (snd_pcm_status_malloc (&status) != 0)) {
949 GST_ERROR_OBJECT (asrc, "snd_pcm_status_malloc failed");
950 return GST_CLOCK_TIME_NONE;
953 if (G_UNLIKELY (snd_pcm_status (asrc->handle, status) != 0)) {
954 GST_ERROR_OBJECT (asrc, "snd_pcm_status failed");
955 return GST_CLOCK_TIME_NONE;
958 /* in case an xrun condition has occurred we need to handle this */
959 if (snd_pcm_status_get_state (status) != SND_PCM_STATE_RUNNING) {
960 if (xrun_recovery (asrc, asrc->handle, err) < 0) {
961 GST_WARNING_OBJECT (asrc, "Could not recover from xrun condition !");
963 /* reload the status alsa status object, since recovery made it invalid */
964 if (G_UNLIKELY (snd_pcm_status (asrc->handle, status) != 0)) {
965 GST_ERROR_OBJECT (asrc, "snd_pcm_status failed");
969 /* get high resolution time stamp from driver */
970 snd_pcm_status_get_htstamp (status, &tstamp);
972 if (tstamp.tv_sec == 0 && tstamp.tv_nsec == 0)
973 return GST_CLOCK_TIME_NONE;
975 timestamp = GST_TIMESPEC_TO_TIME (tstamp);
977 /* max available frames sets the depth of the buffer */
978 avail = snd_pcm_status_get_avail (status);
980 /* calculate the timestamp of the next sample to be read */
981 timestamp -= gst_util_uint64_scale_int (avail, GST_SECOND, asrc->rate);
983 /* compensate for the fact that we really need the timestamp of the
984 * previously read data segment */
985 timestamp -= asrc->period_time * 1000;
987 snd_pcm_status_free (status);
989 GST_LOG_OBJECT (asrc, "ALSA timestamp : %" GST_TIME_FORMAT
990 ", delay %lu", GST_TIME_ARGS (timestamp), avail);
996 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length,
997 GstClockTime * timestamp)
1004 alsa = GST_ALSA_SRC (asrc);
1006 cptr = length / alsa->bpf;
1008 GST_ALSA_SRC_LOCK (asrc);
1010 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) {
1011 if (err == -EAGAIN) {
1012 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err));
1014 } else if (err == -ENODEV) {
1015 goto device_disappeared;
1016 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
1022 ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
1025 GST_ALSA_SRC_UNLOCK (asrc);
1027 /* if driver timestamps are enabled we need to return this here */
1028 if (alsa->driver_timestamps && timestamp)
1029 *timestamp = gst_alsasrc_get_timestamp (alsa);
1031 return length - (cptr * alsa->bpf);
1035 GST_ALSA_SRC_UNLOCK (asrc);
1036 return length; /* skip one period */
1040 GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
1041 (_("Error recording from audio device. "
1042 "The device has been disconnected.")), (NULL));
1043 GST_ALSA_SRC_UNLOCK (asrc);
1049 gst_alsasrc_delay (GstAudioSrc * asrc)
1052 snd_pcm_sframes_t delay;
1055 alsa = GST_ALSA_SRC (asrc);
1057 res = snd_pcm_delay (alsa->handle, &delay);
1058 if (G_UNLIKELY (res < 0)) {
1059 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
1063 return CLAMP (delay, 0, alsa->buffer_size);
1067 gst_alsasrc_reset (GstAudioSrc * asrc)
1072 alsa = GST_ALSA_SRC (asrc);
1074 GST_ALSA_SRC_LOCK (asrc);
1075 GST_DEBUG_OBJECT (alsa, "drop");
1076 CHECK (snd_pcm_drop (alsa->handle), drop_error);
1077 GST_DEBUG_OBJECT (alsa, "prepare");
1078 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
1079 GST_DEBUG_OBJECT (alsa, "reset done");
1080 GST_ALSA_SRC_UNLOCK (asrc);
1087 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
1088 snd_strerror (err));
1089 GST_ALSA_SRC_UNLOCK (asrc);
1094 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
1095 snd_strerror (err));
1096 GST_ALSA_SRC_UNLOCK (asrc);