2 * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
3 * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
20 * Boston, MA 02110-1301, USA.
24 * SECTION:element-alsasink
27 * This element renders raw audio samples using the ALSA audio API.
30 * <title>Example pipelines</title>
32 * gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.ogg ! audioconvert ! audioresample ! autoaudiosink
33 * ]| Play an Ogg/Vorbis file and output audio via ALSA.
40 #include <sys/ioctl.h>
46 #include <alsa/asoundlib.h>
49 #include "gstalsasink.h"
50 #include "gstalsadeviceprobe.h"
52 #include <gst/audio/gstaudioiec61937.h>
53 #include <gst/gst-i18n-plugin.h>
56 #define ESTRPIPE EPIPE
59 #define DEFAULT_DEVICE "default"
60 #define DEFAULT_DEVICE_NAME ""
61 #define DEFAULT_CARD_NAME ""
62 #define SPDIF_PERIOD_SIZE 1536
63 #define SPDIF_BUFFER_SIZE 15360
74 static void gst_alsasink_init_interfaces (GType type);
75 #define gst_alsasink_parent_class parent_class
76 G_DEFINE_TYPE_WITH_CODE (GstAlsaSink, gst_alsasink,
77 GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces (g_define_type_id));
79 static void gst_alsasink_finalise (GObject * object);
80 static void gst_alsasink_set_property (GObject * object,
81 guint prop_id, const GValue * value, GParamSpec * pspec);
82 static void gst_alsasink_get_property (GObject * object,
83 guint prop_id, GValue * value, GParamSpec * pspec);
85 static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink, GstCaps * filter);
86 static gboolean gst_alsasink_query (GstBaseSink * bsink, GstQuery * query);
88 static gboolean gst_alsasink_open (GstAudioSink * asink);
89 static gboolean gst_alsasink_prepare (GstAudioSink * asink,
90 GstAudioRingBufferSpec * spec);
91 static gboolean gst_alsasink_unprepare (GstAudioSink * asink);
92 static gboolean gst_alsasink_close (GstAudioSink * asink);
93 static gint gst_alsasink_write (GstAudioSink * asink, gpointer data,
95 static guint gst_alsasink_delay (GstAudioSink * asink);
96 static void gst_alsasink_reset (GstAudioSink * asink);
97 static gboolean gst_alsasink_acceptcaps (GstAlsaSink * alsa, GstCaps * caps);
98 static GstBuffer *gst_alsasink_payload (GstAudioBaseSink * sink,
101 static gint output_ref; /* 0 */
102 static snd_output_t *output; /* NULL */
103 static GMutex output_mutex;
105 static GstStaticPadTemplate alsasink_sink_factory =
106 GST_STATIC_PAD_TEMPLATE ("sink",
109 GST_STATIC_CAPS ("audio/x-raw, "
110 "format = (string) " GST_AUDIO_FORMATS_ALL ", "
111 "layout = (string) interleaved, "
112 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
117 gst_alsasink_finalise (GObject * object)
119 GstAlsaSink *sink = GST_ALSA_SINK (object);
121 g_free (sink->device);
122 g_mutex_clear (&sink->alsa_lock);
123 g_mutex_clear (&sink->delay_lock);
125 g_mutex_lock (&output_mutex);
127 if (output_ref == 0) {
128 snd_output_close (output);
131 g_mutex_unlock (&output_mutex);
133 G_OBJECT_CLASS (parent_class)->finalize (object);
137 gst_alsasink_init_interfaces (GType type)
140 gst_alsa_type_add_device_property_probe_interface (type);
145 gst_alsasink_class_init (GstAlsaSinkClass * klass)
147 GObjectClass *gobject_class;
148 GstElementClass *gstelement_class;
149 GstBaseSinkClass *gstbasesink_class;
150 GstAudioBaseSinkClass *gstbaseaudiosink_class;
151 GstAudioSinkClass *gstaudiosink_class;
153 gobject_class = (GObjectClass *) klass;
154 gstelement_class = (GstElementClass *) klass;
155 gstbasesink_class = (GstBaseSinkClass *) klass;
156 gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;
157 gstaudiosink_class = (GstAudioSinkClass *) klass;
159 parent_class = g_type_class_peek_parent (klass);
161 gobject_class->finalize = gst_alsasink_finalise;
162 gobject_class->get_property = gst_alsasink_get_property;
163 gobject_class->set_property = gst_alsasink_set_property;
165 gst_element_class_set_static_metadata (gstelement_class,
166 "Audio sink (ALSA)", "Sink/Audio",
167 "Output to a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
169 gst_element_class_add_pad_template (gstelement_class,
170 gst_static_pad_template_get (&alsasink_sink_factory));
172 gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);
173 gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_alsasink_query);
175 gstbaseaudiosink_class->payload = GST_DEBUG_FUNCPTR (gst_alsasink_payload);
177 gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open);
178 gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare);
179 gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare);
180 gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close);
181 gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write);
182 gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay);
183 gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink_reset);
185 g_object_class_install_property (gobject_class, PROP_DEVICE,
186 g_param_spec_string ("device", "Device",
187 "ALSA device, as defined in an asound configuration file",
188 DEFAULT_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
190 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
191 g_param_spec_string ("device-name", "Device name",
192 "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
193 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
195 g_object_class_install_property (gobject_class, PROP_CARD_NAME,
196 g_param_spec_string ("card-name", "Card name",
197 "Human-readable name of the sound card", DEFAULT_CARD_NAME,
198 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
202 gst_alsasink_set_property (GObject * object, guint prop_id,
203 const GValue * value, GParamSpec * pspec)
207 sink = GST_ALSA_SINK (object);
211 g_free (sink->device);
212 sink->device = g_value_dup_string (value);
213 /* setting NULL restores the default device */
214 if (sink->device == NULL) {
215 sink->device = g_strdup (DEFAULT_DEVICE);
219 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
225 gst_alsasink_get_property (GObject * object, guint prop_id,
226 GValue * value, GParamSpec * pspec)
230 sink = GST_ALSA_SINK (object);
234 g_value_set_string (value, sink->device);
236 case PROP_DEVICE_NAME:
237 g_value_take_string (value,
238 gst_alsa_find_device_name (GST_OBJECT_CAST (sink),
239 sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK));
242 g_value_take_string (value,
243 gst_alsa_find_card_name (GST_OBJECT_CAST (sink),
244 sink->device, SND_PCM_STREAM_PLAYBACK));
247 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
253 gst_alsasink_init (GstAlsaSink * alsasink)
255 GST_DEBUG_OBJECT (alsasink, "initializing alsasink");
257 alsasink->device = g_strdup (DEFAULT_DEVICE);
258 alsasink->handle = NULL;
259 alsasink->cached_caps = NULL;
260 g_mutex_init (&alsasink->alsa_lock);
261 g_mutex_init (&alsasink->delay_lock);
263 g_mutex_lock (&output_mutex);
264 if (output_ref == 0) {
265 snd_output_stdio_attach (&output, stdout, 0);
268 g_mutex_unlock (&output_mutex);
271 #define CHECK(call, error) \
273 if ((err = call) < 0) { \
274 GST_WARNING_OBJECT (alsa, "Error %d (%s) calling " #call, err, snd_strerror (err)); \
280 gst_alsasink_getcaps (GstBaseSink * bsink, GstCaps * filter)
282 GstElementClass *element_class;
283 GstPadTemplate *pad_template;
284 GstAlsaSink *sink = GST_ALSA_SINK (bsink);
285 GstCaps *caps, *templ_caps;
287 GST_OBJECT_LOCK (sink);
288 if (sink->handle == NULL) {
289 GST_OBJECT_UNLOCK (sink);
290 GST_DEBUG_OBJECT (sink, "device not open, using template caps");
291 return NULL; /* base class will get template caps for us */
294 if (sink->cached_caps) {
296 caps = gst_caps_intersect_full (filter, sink->cached_caps,
297 GST_CAPS_INTERSECT_FIRST);
298 GST_OBJECT_UNLOCK (sink);
299 GST_LOG_OBJECT (sink, "Returning cached caps %" GST_PTR_FORMAT " with "
300 "filter %" GST_PTR_FORMAT " applied: %" GST_PTR_FORMAT,
301 sink->cached_caps, filter, caps);
304 caps = gst_caps_ref (sink->cached_caps);
305 GST_OBJECT_UNLOCK (sink);
306 GST_LOG_OBJECT (sink, "Returning cached caps %" GST_PTR_FORMAT, caps);
311 element_class = GST_ELEMENT_GET_CLASS (sink);
312 pad_template = gst_element_class_get_pad_template (element_class, "sink");
313 if (pad_template == NULL) {
314 GST_OBJECT_UNLOCK (sink);
315 g_assert_not_reached ();
319 templ_caps = gst_pad_template_get_caps (pad_template);
320 caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->device,
321 sink->handle, templ_caps);
322 gst_caps_unref (templ_caps);
325 sink->cached_caps = gst_caps_ref (caps);
328 GST_OBJECT_UNLOCK (sink);
330 GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);
333 GstCaps *intersection;
336 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
337 gst_caps_unref (caps);
345 gst_alsasink_acceptcaps (GstAlsaSink * alsa, GstCaps * caps)
347 GstPad *pad = GST_BASE_SINK (alsa)->sinkpad;
350 gboolean ret = FALSE;
351 GstAudioRingBufferSpec spec = { 0 };
353 pad_caps = gst_pad_query_caps (pad, caps);
354 if (!pad_caps || gst_caps_is_empty (pad_caps)) {
356 gst_caps_unref (pad_caps);
360 gst_caps_unref (pad_caps);
362 /* If we've not got fixed caps, creating a stream might fail, so let's just
363 * return from here with default acceptcaps behaviour */
364 if (!gst_caps_is_fixed (caps))
367 /* parse helper expects this set, so avoid nasty warning
368 * will be set properly later on anyway */
369 spec.latency_time = GST_SECOND;
370 if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
373 /* Make sure input is framed (one frame per buffer) and can be payloaded */
375 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
376 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
377 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
378 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
380 gboolean framed = FALSE, parsed = FALSE;
381 st = gst_caps_get_structure (caps, 0);
383 gst_structure_get_boolean (st, "framed", &framed);
384 gst_structure_get_boolean (st, "parsed", &parsed);
385 if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
394 gst_caps_replace (&spec.caps, NULL);
399 gst_alsasink_query (GstBaseSink * sink, GstQuery * query)
401 GstAlsaSink *alsa = GST_ALSA_SINK (sink);
404 switch (GST_QUERY_TYPE (query)) {
405 case GST_QUERY_ACCEPT_CAPS:
409 gst_query_parse_accept_caps (query, &caps);
410 ret = gst_alsasink_acceptcaps (alsa, caps);
411 gst_query_set_accept_caps_result (query, ret);
416 ret = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
423 set_hwparams (GstAlsaSink * alsa)
427 snd_pcm_hw_params_t *params;
428 guint period_time, buffer_time;
430 snd_pcm_hw_params_malloc (¶ms);
432 GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "
433 "SPDIF (%d)", alsa->channels, alsa->rate,
434 snd_pcm_format_name (alsa->format), alsa->iec958);
436 /* start with requested values, if we cannot configure alsa for those values,
437 * we set these values to -1, which will leave the default alsa values */
438 buffer_time = alsa->buffer_time;
439 period_time = alsa->period_time;
442 /* choose all parameters */
443 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
444 /* set the interleaved read/write format */
445 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
447 /* set the sample format */
449 /* Try to use big endian first else fallback to le and swap bytes */
450 if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {
451 alsa->format = SND_PCM_FORMAT_S16_LE;
452 alsa->need_swap = TRUE;
453 GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");
455 alsa->need_swap = FALSE;
458 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
460 /* set the count of channels */
461 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
463 /* set the stream rate */
465 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
468 #ifndef GST_DISABLE_GST_DEBUG
469 /* get and dump some limits */
473 snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL);
474 snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);
476 GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",
477 alsa->buffer_time, min, max);
479 snd_pcm_hw_params_get_period_time_min (params, &min, NULL);
480 snd_pcm_hw_params_get_period_time_max (params, &max, NULL);
482 GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",
483 alsa->period_time, min, max);
485 snd_pcm_hw_params_get_periods_min (params, &min, NULL);
486 snd_pcm_hw_params_get_periods_max (params, &max, NULL);
488 GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
492 /* now try to configure the buffer time and period time, if one
493 * of those fail, we fall back to the defaults and emit a warning. */
494 if (buffer_time != -1 && !alsa->iec958) {
495 /* set the buffer time */
496 if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
497 &buffer_time, NULL)) < 0) {
498 GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
499 ("Unable to set buffer time %i for playback: %s",
500 buffer_time, snd_strerror (err)));
501 /* disable buffer_time the next round */
505 GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time);
506 alsa->buffer_time = buffer_time;
508 if (period_time != -1 && !alsa->iec958) {
509 /* set the period time */
510 if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
511 &period_time, NULL)) < 0) {
512 GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
513 ("Unable to set period time %i for playback: %s",
514 period_time, snd_strerror (err)));
515 /* disable period_time the next round */
519 GST_DEBUG_OBJECT (alsa, "period time %u", period_time);
520 alsa->period_time = period_time;
523 /* Set buffer size and period size manually for SPDIF */
524 if (G_UNLIKELY (alsa->iec958)) {
525 snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;
526 snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;
528 CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,
529 &buffer_size), buffer_size);
530 CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,
531 &period_size, NULL), period_size);
534 /* write the parameters to device */
535 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
537 /* now get the configured values */
538 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
540 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
543 GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
546 snd_pcm_hw_params_free (params);
552 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
553 ("Broken configuration for playback: no configurations available: %s",
554 snd_strerror (err)));
555 snd_pcm_hw_params_free (params);
560 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
561 ("Access type not available for playback: %s", snd_strerror (err)));
562 snd_pcm_hw_params_free (params);
567 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
568 ("Sample format not available for playback: %s", snd_strerror (err)));
569 snd_pcm_hw_params_free (params);
576 if ((alsa->channels) == 1)
577 msg = g_strdup (_("Could not open device for playback in mono mode."));
578 if ((alsa->channels) == 2)
579 msg = g_strdup (_("Could not open device for playback in stereo mode."));
580 if ((alsa->channels) > 2)
583 ("Could not open device for playback in %d-channel mode."),
585 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
586 ("%s", snd_strerror (err)));
588 snd_pcm_hw_params_free (params);
593 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
594 ("Rate %iHz not available for playback: %s",
595 alsa->rate, snd_strerror (err)));
600 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
601 ("Unable to get buffer size for playback: %s", snd_strerror (err)));
602 snd_pcm_hw_params_free (params);
607 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
608 ("Unable to get period size for playback: %s", snd_strerror (err)));
609 snd_pcm_hw_params_free (params);
614 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
615 ("Unable to set hw params for playback: %s", snd_strerror (err)));
616 snd_pcm_hw_params_free (params);
622 set_swparams (GstAlsaSink * alsa)
625 snd_pcm_sw_params_t *params;
627 snd_pcm_sw_params_malloc (¶ms);
629 /* get the current swparams */
630 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
631 /* start the transfer when the buffer is almost full: */
632 /* (buffer_size / avail_min) * avail_min */
633 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
634 (alsa->buffer_size / alsa->period_size) * alsa->period_size),
637 /* allow the transfer when at least period_size samples can be processed */
638 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
639 alsa->period_size), set_avail);
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 playback 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 alsasink_parse_spec (GstAlsaSink * alsa, GstAudioRingBufferSpec * spec)
699 /* Initialize our boolean */
700 alsa->iec958 = FALSE;
702 switch (spec->type) {
703 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
704 switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
705 case GST_AUDIO_FORMAT_U8:
706 alsa->format = SND_PCM_FORMAT_U8;
708 case GST_AUDIO_FORMAT_S8:
709 alsa->format = SND_PCM_FORMAT_S8;
711 case GST_AUDIO_FORMAT_S16LE:
712 alsa->format = SND_PCM_FORMAT_S16_LE;
714 case GST_AUDIO_FORMAT_S16BE:
715 alsa->format = SND_PCM_FORMAT_S16_BE;
717 case GST_AUDIO_FORMAT_U16LE:
718 alsa->format = SND_PCM_FORMAT_U16_LE;
720 case GST_AUDIO_FORMAT_U16BE:
721 alsa->format = SND_PCM_FORMAT_U16_BE;
723 case GST_AUDIO_FORMAT_S24_32LE:
724 alsa->format = SND_PCM_FORMAT_S24_LE;
726 case GST_AUDIO_FORMAT_S24_32BE:
727 alsa->format = SND_PCM_FORMAT_S24_BE;
729 case GST_AUDIO_FORMAT_U24_32LE:
730 alsa->format = SND_PCM_FORMAT_U24_LE;
732 case GST_AUDIO_FORMAT_U24_32BE:
733 alsa->format = SND_PCM_FORMAT_U24_BE;
735 case GST_AUDIO_FORMAT_S32LE:
736 alsa->format = SND_PCM_FORMAT_S32_LE;
738 case GST_AUDIO_FORMAT_S32BE:
739 alsa->format = SND_PCM_FORMAT_S32_BE;
741 case GST_AUDIO_FORMAT_U32LE:
742 alsa->format = SND_PCM_FORMAT_U32_LE;
744 case GST_AUDIO_FORMAT_U32BE:
745 alsa->format = SND_PCM_FORMAT_U32_BE;
747 case GST_AUDIO_FORMAT_S24LE:
748 alsa->format = SND_PCM_FORMAT_S24_3LE;
750 case GST_AUDIO_FORMAT_S24BE:
751 alsa->format = SND_PCM_FORMAT_S24_3BE;
753 case GST_AUDIO_FORMAT_U24LE:
754 alsa->format = SND_PCM_FORMAT_U24_3LE;
756 case GST_AUDIO_FORMAT_U24BE:
757 alsa->format = SND_PCM_FORMAT_U24_3BE;
759 case GST_AUDIO_FORMAT_S20LE:
760 alsa->format = SND_PCM_FORMAT_S20_3LE;
762 case GST_AUDIO_FORMAT_S20BE:
763 alsa->format = SND_PCM_FORMAT_S20_3BE;
765 case GST_AUDIO_FORMAT_U20LE:
766 alsa->format = SND_PCM_FORMAT_U20_3LE;
768 case GST_AUDIO_FORMAT_U20BE:
769 alsa->format = SND_PCM_FORMAT_U20_3BE;
771 case GST_AUDIO_FORMAT_S18LE:
772 alsa->format = SND_PCM_FORMAT_S18_3LE;
774 case GST_AUDIO_FORMAT_S18BE:
775 alsa->format = SND_PCM_FORMAT_S18_3BE;
777 case GST_AUDIO_FORMAT_U18LE:
778 alsa->format = SND_PCM_FORMAT_U18_3LE;
780 case GST_AUDIO_FORMAT_U18BE:
781 alsa->format = SND_PCM_FORMAT_U18_3BE;
783 case GST_AUDIO_FORMAT_F32LE:
784 alsa->format = SND_PCM_FORMAT_FLOAT_LE;
786 case GST_AUDIO_FORMAT_F32BE:
787 alsa->format = SND_PCM_FORMAT_FLOAT_BE;
789 case GST_AUDIO_FORMAT_F64LE:
790 alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
792 case GST_AUDIO_FORMAT_F64BE:
793 alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
799 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
800 alsa->format = SND_PCM_FORMAT_A_LAW;
802 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
803 alsa->format = SND_PCM_FORMAT_MU_LAW;
805 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
806 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
807 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
808 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
809 alsa->format = SND_PCM_FORMAT_S16_BE;
816 alsa->rate = GST_AUDIO_INFO_RATE (&spec->info);
817 alsa->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
818 alsa->buffer_time = spec->buffer_time;
819 alsa->period_time = spec->latency_time;
820 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
822 if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9)
823 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK
824 (alsa)->ringbuffer, alsa_position[alsa->channels - 1]);
836 gst_alsasink_open (GstAudioSink * asink)
841 alsa = GST_ALSA_SINK (asink);
843 /* open in non-blocking mode, we'll use snd_pcm_wait() for space to become
845 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,
846 SND_PCM_NONBLOCK), open_error);
847 GST_LOG_OBJECT (alsa, "Opened device %s", alsa->device);
855 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
856 (_("Could not open audio device for playback. "
857 "Device is being used by another application.")),
858 ("Device '%s' is busy", alsa->device));
860 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,
861 (_("Could not open audio device for playback.")),
862 ("Playback open error on device '%s': %s", alsa->device,
863 snd_strerror (err)));
870 gst_alsasink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
875 alsa = GST_ALSA_SINK (asink);
878 snd_pcm_close (alsa->handle);
879 alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa), alsa->device);
880 if (G_UNLIKELY (!alsa->handle)) {
885 if (!alsasink_parse_spec (alsa, spec))
888 CHECK (set_hwparams (alsa), hw_params_failed);
889 CHECK (set_swparams (alsa), sw_params_failed);
891 alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
892 spec->segsize = alsa->period_size * alsa->bpf;
893 spec->segtotal = alsa->buffer_size / alsa->period_size;
896 snd_output_t *out_buf = NULL;
899 snd_output_buffer_open (&out_buf);
900 snd_pcm_dump_hw_setup (alsa->handle, out_buf);
901 snd_output_buffer_string (out_buf, &msg);
902 GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
903 snd_output_close (out_buf);
904 snd_output_buffer_open (&out_buf);
905 snd_pcm_dump_sw_setup (alsa->handle, out_buf);
906 snd_output_buffer_string (out_buf, &msg);
907 GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
908 snd_output_close (out_buf);
911 #ifdef SND_CHMAP_API_VERSION
912 if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9) {
913 snd_pcm_chmap_t *chmap = snd_pcm_get_chmap (alsa->handle);
914 if (chmap && chmap->channels == alsa->channels) {
915 GstAudioChannelPosition pos[8];
916 if (alsa_chmap_to_channel_positions (chmap, pos))
917 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK
918 (alsa)->ringbuffer, pos);
922 #endif /* SND_CHMAP_API_VERSION */
929 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL),
930 ("Could not open IEC958 (SPDIF) device for playback"));
935 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
936 ("Error parsing spec"));
941 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
942 ("Setting of hwparams failed: %s", snd_strerror (err)));
947 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
948 ("Setting of swparams failed: %s", snd_strerror (err)));
954 gst_alsasink_unprepare (GstAudioSink * asink)
958 alsa = GST_ALSA_SINK (asink);
960 snd_pcm_drop (alsa->handle);
961 snd_pcm_hw_free (alsa->handle);
967 gst_alsasink_close (GstAudioSink * asink)
969 GstAlsaSink *alsa = GST_ALSA_SINK (asink);
971 GST_OBJECT_LOCK (asink);
973 snd_pcm_close (alsa->handle);
976 gst_caps_replace (&alsa->cached_caps, NULL);
977 GST_OBJECT_UNLOCK (asink);
984 * Underrun and suspend recovery
987 xrun_recovery (GstAlsaSink * alsa, snd_pcm_t * handle, gint err)
989 GST_WARNING_OBJECT (alsa, "xrun recovery %d: %s", err, g_strerror (-err));
991 if (err == -EPIPE) { /* under-run */
992 err = snd_pcm_prepare (handle);
994 GST_WARNING_OBJECT (alsa,
995 "Can't recover from underrun, prepare failed: %s",
997 gst_audio_base_sink_report_device_failure (GST_AUDIO_BASE_SINK (alsa));
999 } else if (err == -ESTRPIPE) {
1000 while ((err = snd_pcm_resume (handle)) == -EAGAIN)
1001 g_usleep (100); /* wait until the suspend flag is released */
1004 err = snd_pcm_prepare (handle);
1006 GST_WARNING_OBJECT (alsa,
1007 "Can't recover from suspend, prepare failed: %s",
1008 snd_strerror (err));
1011 gst_audio_base_sink_report_device_failure (GST_AUDIO_BASE_SINK (alsa));
1018 gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
1025 alsa = GST_ALSA_SINK (asink);
1027 if (alsa->iec958 && alsa->need_swap) {
1029 guint16 *ptr_tmp = (guint16 *) ptr;
1031 GST_DEBUG_OBJECT (asink, "swapping bytes");
1032 for (i = 0; i < length / 2; i++) {
1033 ptr_tmp[i] = GUINT16_SWAP_LE_BE (ptr_tmp[i]);
1037 GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length);
1039 cptr = length / alsa->bpf;
1041 GST_ALSA_SINK_LOCK (asink);
1043 /* start by doing a blocking wait for free space. Set the timeout
1044 * to 4 times the period time */
1045 err = snd_pcm_wait (alsa->handle, (4 * alsa->period_time / 1000));
1047 GST_DEBUG_OBJECT (asink, "wait error, %d", err);
1049 GST_DELAY_SINK_LOCK (asink);
1050 err = snd_pcm_writei (alsa->handle, ptr, cptr);
1051 GST_DELAY_SINK_UNLOCK (asink);
1054 GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr);
1056 GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err));
1057 if (err == -EAGAIN) {
1059 } else if (err == -ENODEV) {
1060 goto device_disappeared;
1061 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
1067 ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
1070 GST_ALSA_SINK_UNLOCK (asink);
1072 return length - (cptr * alsa->bpf);
1076 GST_ALSA_SINK_UNLOCK (asink);
1077 return length; /* skip one period */
1081 GST_ELEMENT_ERROR (asink, RESOURCE, WRITE,
1082 (_("Error outputting to audio device. "
1083 "The device has been disconnected.")), (NULL));
1089 gst_alsasink_delay (GstAudioSink * asink)
1092 snd_pcm_sframes_t delay;
1095 alsa = GST_ALSA_SINK (asink);
1097 GST_DELAY_SINK_LOCK (asink);
1098 res = snd_pcm_delay (alsa->handle, &delay);
1099 GST_DELAY_SINK_UNLOCK (asink);
1100 if (G_UNLIKELY (res < 0)) {
1101 /* on errors, report 0 delay */
1102 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
1105 if (G_UNLIKELY (delay < 0)) {
1106 /* make sure we never return a negative delay */
1107 GST_WARNING_OBJECT (alsa, "snd_pcm_delay returned negative delay");
1115 gst_alsasink_reset (GstAudioSink * asink)
1120 alsa = GST_ALSA_SINK (asink);
1122 GST_ALSA_SINK_LOCK (asink);
1123 GST_DEBUG_OBJECT (alsa, "drop");
1124 CHECK (snd_pcm_drop (alsa->handle), drop_error);
1125 GST_DEBUG_OBJECT (alsa, "prepare");
1126 CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
1127 GST_DEBUG_OBJECT (alsa, "reset done");
1128 GST_ALSA_SINK_UNLOCK (asink);
1135 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
1136 snd_strerror (err));
1137 GST_ALSA_SINK_UNLOCK (asink);
1142 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
1143 snd_strerror (err));
1144 GST_ALSA_SINK_UNLOCK (asink);
1150 gst_alsasink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
1154 alsa = GST_ALSA_SINK (sink);
1159 GstMapInfo iinfo, oinfo;
1161 framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
1165 out = gst_buffer_new_and_alloc (framesize);
1167 gst_buffer_map (buf, &iinfo, GST_MAP_READ);
1168 gst_buffer_map (out, &oinfo, GST_MAP_WRITE);
1170 if (!gst_audio_iec61937_payload (iinfo.data, iinfo.size,
1171 oinfo.data, oinfo.size, &sink->ringbuffer->spec, G_BIG_ENDIAN)) {
1172 gst_buffer_unref (out);
1176 gst_buffer_unmap (buf, &iinfo);
1177 gst_buffer_unmap (out, &oinfo);
1179 gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_METADATA, 0, -1);
1183 return gst_buffer_ref (buf);