1 /* GStreamer

     2  * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>

     3  * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>

     4  *

     5  * gstalsasink.c:

     6  *

     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.

    11  *

    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.

    16  *

    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., 59 Temple Place - Suite 330,

    20  * Boston, MA 02111-1307, USA.

    21  */

    22 

    23 /**

    24  * SECTION:element-alsasink

    25  * @short_description: play audio to an ALSA device

    26  * @see_also: alsasrc, alsamixer

    27  *

    28  * <refsect2>

    29  * <para>

    30  * This element renders raw audio samples using the ALSA api.

    31  * </para>

    32  * <title>Example pipelines</title>

    33  * <para>

    34  * Play an Ogg/Vorbis file.

    35  * </para>

    36  * <programlisting>

    37  * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink

    38  * </programlisting>

    39  * </refsect2>

    40  *

    41  * Last reviewed on 2006-03-01 (0.10.4)

    42  */

    43 

    44 #ifdef HAVE_CONFIG_H

    45 #include "config.h"

    46 #endif

    47 #include <sys/ioctl.h>

    48 #include <fcntl.h>

    49 #include <errno.h>

    50 #include <unistd.h>

    51 #include <string.h>

    52 #include <getopt.h>

    53 #include <alsa/asoundlib.h>

    54 

    55 #include "gstalsa.h"

    56 #include "gstalsasink.h"

    57 #include "gstalsadeviceprobe.h"

    58 

    59 #include <gst/gst-i18n-plugin.h>

    60 

    61 /* elementfactory information */

    62 static const GstElementDetails gst_alsasink_details =

    63 GST_ELEMENT_DETAILS ("Audio sink (ALSA)",

    64     "Sink/Audio",

    65     "Output to a sound card via ALSA",

    66     "Wim Taymans <wim@fluendo.com>");

    67 

    68 #define DEFAULT_DEVICE		"default"

    69 #define DEFAULT_DEVICE_NAME	""

    70 #define SPDIF_PERIOD_SIZE 1536

    71 #define SPDIF_BUFFER_SIZE 15360

    72 

    73 enum

    74 {

    75   PROP_0,

    76   PROP_DEVICE,

    77   PROP_DEVICE_NAME

    78 };

    79 

    80 static void gst_alsasink_init_interfaces (GType type);

    81 

    82 GST_BOILERPLATE_FULL (GstAlsaSink, gst_alsasink, GstAudioSink,

    83     GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces);

    84 

    85 static void gst_alsasink_finalise (GObject * object);

    86 static void gst_alsasink_set_property (GObject * object,

    87     guint prop_id, const GValue * value, GParamSpec * pspec);

    88 static void gst_alsasink_get_property (GObject * object,

    89     guint prop_id, GValue * value, GParamSpec * pspec);

    90 

    91 static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink);

    92 

    93 static gboolean gst_alsasink_open (GstAudioSink * asink);

    94 static gboolean gst_alsasink_prepare (GstAudioSink * asink,

    95     GstRingBufferSpec * spec);

    96 static gboolean gst_alsasink_unprepare (GstAudioSink * asink);

    97 static gboolean gst_alsasink_close (GstAudioSink * asink);

    98 static guint gst_alsasink_write (GstAudioSink * asink, gpointer data,

    99     guint length);

   100 static guint gst_alsasink_delay (GstAudioSink * asink);

   101 static void gst_alsasink_reset (GstAudioSink * asink);

   102 

   103 static gint output_ref;         /* 0    */

   104 static snd_output_t *output;    /* NULL */

   105 static GStaticMutex output_mutex = G_STATIC_MUTEX_INIT;

   106 

   107 

   108 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN)

   109 # define ALSA_SINK_FACTORY_ENDIANNESS	"LITTLE_ENDIAN, BIG_ENDIAN"

   110 #else

   111 # define ALSA_SINK_FACTORY_ENDIANNESS	"BIG_ENDIAN, LITTLE_ENDIAN"

   112 #endif

   113 

   114 static GstStaticPadTemplate alsasink_sink_factory =

   115     GST_STATIC_PAD_TEMPLATE ("sink",

   116     GST_PAD_SINK,

   117     GST_PAD_ALWAYS,

   118     GST_STATIC_CAPS ("audio/x-raw-int, "

   119         "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "

   120         "signed = (boolean) { TRUE, FALSE }, "

   121         "width = (int) 32, "

   122         "depth = (int) 32, "

   123         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "

   124         "audio/x-raw-int, "

   125         "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "

   126         "signed = (boolean) { TRUE, FALSE }, "

   127         "width = (int) 24, "

   128         "depth = (int) 24, "

   129         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "

   130         "audio/x-raw-int, "

   131         "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "

   132         "signed = (boolean) { TRUE, FALSE }, "

   133         "width = (int) 32, "

   134         "depth = (int) 24, "

   135         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "

   136         "audio/x-raw-int, "

   137         "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "

   138         "signed = (boolean) { TRUE, FALSE }, "

   139         "width = (int) 16, "

   140         "depth = (int) 16, "

   141         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "

   142         "audio/x-raw-int, "

   143         "signed = (boolean) { TRUE, FALSE }, "

   144         "width = (int) 8, "

   145         "depth = (int) 8, "

   146         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ];"

   147         "audio/x-iec958")

   148     );

   149 

   150 static void

   151 gst_alsasink_finalise (GObject * object)

   152 {

   153   GstAlsaSink *sink = GST_ALSA_SINK (object);

   154 

   155   g_free (sink->device);

   156   g_mutex_free (sink->alsa_lock);

   157 

   158   g_static_mutex_lock (&output_mutex);

   159   --output_ref;

   160   if (output_ref == 0) {

   161     snd_output_close (output);

   162     output = NULL;

   163   }

   164   g_static_mutex_unlock (&output_mutex);

   165 

   166   G_OBJECT_CLASS (parent_class)->finalize (object);

   167 }

   168 

   169 static void

   170 gst_alsasink_init_interfaces (GType type)

   171 {

   172   gst_alsa_type_add_device_property_probe_interface (type);

   173 }

   174 

   175 static void

   176 gst_alsasink_base_init (gpointer g_class)

   177 {

   178   GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

   179 

   180   gst_element_class_set_details (element_class, &gst_alsasink_details);

   181 

   182   gst_element_class_add_pad_template (element_class,

   183       gst_static_pad_template_get (&alsasink_sink_factory));

   184 }

   185 static void

   186 gst_alsasink_class_init (GstAlsaSinkClass * klass)

   187 {

   188   GObjectClass *gobject_class;

   189   GstElementClass *gstelement_class;

   190   GstBaseSinkClass *gstbasesink_class;

   191   GstBaseAudioSinkClass *gstbaseaudiosink_class;

   192   GstAudioSinkClass *gstaudiosink_class;

   193 

   194   gobject_class = (GObjectClass *) klass;

   195   gstelement_class = (GstElementClass *) klass;

   196   gstbasesink_class = (GstBaseSinkClass *) klass;

   197   gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;

   198   gstaudiosink_class = (GstAudioSinkClass *) klass;

   199 

   200   parent_class = g_type_class_peek_parent (klass);

   201 

   202   gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasink_finalise);

   203   gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasink_get_property);

   204   gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasink_set_property);

   205 

   206   gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);

   207 

   208   gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open);

   209   gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare);

   210   gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare);

   211   gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close);

   212   gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write);

   213   gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay);

   214   gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink_reset);

   215 

   216   g_object_class_install_property (gobject_class, PROP_DEVICE,

   217       g_param_spec_string ("device", "Device",

   218           "ALSA device, as defined in an asound configuration file",

   219           DEFAULT_DEVICE, G_PARAM_READWRITE));

   220 

   221   g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,

   222       g_param_spec_string ("device-name", "Device name",

   223           "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,

   224           G_PARAM_READABLE));

   225 }

   226 

   227 static void

   228 gst_alsasink_set_property (GObject * object, guint prop_id,

   229     const GValue * value, GParamSpec * pspec)

   230 {

   231   GstAlsaSink *sink;

   232 

   233   sink = GST_ALSA_SINK (object);

   234 

   235   switch (prop_id) {

   236     case PROP_DEVICE:

   237       g_free (sink->device);

   238       sink->device = g_value_dup_string (value);

   239       /* setting NULL restores the default device */

   240       if (sink->device == NULL) {

   241         sink->device = g_strdup (DEFAULT_DEVICE);

   242       }

   243       break;

   244     default:

   245       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

   246       break;

   247   }

   248 }

   249 

   250 static void

   251 gst_alsasink_get_property (GObject * object, guint prop_id,

   252     GValue * value, GParamSpec * pspec)

   253 {

   254   GstAlsaSink *sink;

   255 

   256   sink = GST_ALSA_SINK (object);

   257 

   258   switch (prop_id) {

   259     case PROP_DEVICE:

   260       g_value_set_string (value, sink->device);

   261       break;

   262     case PROP_DEVICE_NAME:

   263       g_value_take_string (value,

   264           gst_alsa_find_device_name (GST_OBJECT_CAST (sink),

   265               sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK));

   266       break;

   267     default:

   268       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

   269       break;

   270   }

   271 }

   272 

   273 static void

   274 gst_alsasink_init (GstAlsaSink * alsasink, GstAlsaSinkClass * g_class)

   275 {

   276   GST_DEBUG_OBJECT (alsasink, "initializing alsasink");

   277 

   278   alsasink->device = g_strdup (DEFAULT_DEVICE);

   279   alsasink->handle = NULL;

   280   alsasink->cached_caps = NULL;

   281   alsasink->alsa_lock = g_mutex_new ();

   282 

   283   g_static_mutex_lock (&output_mutex);

   284   if (output_ref == 0) {

   285     snd_output_stdio_attach (&output, stdout, 0);

   286     ++output_ref;

   287   }

   288   g_static_mutex_unlock (&output_mutex);

   289 }

   290 

   291 #define CHECK(call, error) \

   292 G_STMT_START {                  \

   293 if ((err = call) < 0)           \

   294   goto error;                   \

   295 } G_STMT_END;

   296 

   297 static GstCaps *

   298 gst_alsasink_getcaps (GstBaseSink * bsink)

   299 {

   300   GstElementClass *element_class;

   301   GstPadTemplate *pad_template;

   302   GstAlsaSink *sink = GST_ALSA_SINK (bsink);

   303   GstCaps *caps;

   304 

   305   if (sink->handle == NULL) {

   306     GST_DEBUG_OBJECT (sink, "device not open, using template caps");

   307     return NULL;                /* base class will get template caps for us */

   308   }

   309 

   310   if (sink->cached_caps) {

   311     GST_LOG_OBJECT (sink, "Returning cached caps");

   312     return gst_caps_ref (sink->cached_caps);

   313   }

   314 

   315   element_class = GST_ELEMENT_GET_CLASS (sink);

   316   pad_template = gst_element_class_get_pad_template (element_class, "sink");

   317   g_return_val_if_fail (pad_template != NULL, NULL);

   318 

   319   caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->handle,

   320       gst_pad_template_get_caps (pad_template));

   321 

   322   if (caps) {

   323     sink->cached_caps = gst_caps_ref (caps);

   324   }

   325 

   326   GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);

   327 

   328   return caps;

   329 }

   330 

   331 static int

   332 set_hwparams (GstAlsaSink * alsa)

   333 {

   334   guint rrate;

   335   gint err, dir;

   336   snd_pcm_hw_params_t *params;

   337   guint period_time, buffer_time;

   338 

   339   snd_pcm_hw_params_malloc (&params);

   340 

   341   GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "

   342       "SPDIF (%d)", alsa->channels, alsa->rate,

   343       snd_pcm_format_name (alsa->format), alsa->iec958);

   344 

   345   /* start with requested values, if we cannot configure alsa for those values,

   346    * we set these values to -1, which will leave the default alsa values */

   347   buffer_time = alsa->buffer_time;

   348   period_time = alsa->period_time;

   349 

   350 retry:

   351   /* choose all parameters */

   352   CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);

   353   /* set the interleaved read/write format */

   354   CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),

   355       wrong_access);

   356   /* set the sample format */

   357   if (alsa->iec958) {

   358     /* Try to use big endian first else fallback to le and swap bytes */

   359     if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {

   360       alsa->format = SND_PCM_FORMAT_S16_LE;

   361       alsa->need_swap = TRUE;

   362       GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");

   363     } else {

   364       alsa->need_swap = FALSE;

   365     }

   366   }

   367   CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),

   368       no_sample_format);

   369   /* set the count of channels */

   370   CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),

   371       no_channels);

   372   /* set the stream rate */

   373   rrate = alsa->rate;

   374   CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),

   375       no_rate);

   376   if (rrate != alsa->rate)

   377     goto rate_match;

   378 

   379   /* get and dump some limits */

   380   {

   381     guint min, max;

   382 

   383     snd_pcm_hw_params_get_buffer_time_min (params, &min, &dir);

   384     snd_pcm_hw_params_get_buffer_time_max (params, &max, &dir);

   385 

   386     GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",

   387         alsa->buffer_time, min, max);

   388 

   389     snd_pcm_hw_params_get_period_time_min (params, &min, &dir);

   390     snd_pcm_hw_params_get_period_time_max (params, &max, &dir);

   391 

   392     GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",

   393         alsa->period_time, min, max);

   394 

   395     snd_pcm_hw_params_get_periods_min (params, &min, &dir);

   396     snd_pcm_hw_params_get_periods_max (params, &max, &dir);

   397 

   398     GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);

   399   }

   400 

   401   /* now try to configure the buffer time and period time, if one

   402    * of those fail, we fall back to the defaults and emit a warning. */

   403   if (buffer_time != -1 && !alsa->iec958) {

   404     /* set the buffer time */

   405     if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,

   406                 &buffer_time, &dir)) < 0) {

   407       GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),

   408           ("Unable to set buffer time %i for playback: %s",

   409               buffer_time, snd_strerror (err)));

   410       /* disable buffer_time the next round */

   411       buffer_time = -1;

   412       goto retry;

   413     }

   414     GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time);

   415   }

   416   if (period_time != -1 && !alsa->iec958) {

   417     /* set the period time */

   418     if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params,

   419                 &period_time, &dir)) < 0) {

   420       GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),

   421           ("Unable to set period time %i for playback: %s",

   422               period_time, snd_strerror (err)));

   423       /* disable period_time the next round */

   424       period_time = -1;

   425       goto retry;

   426     }

   427     GST_DEBUG_OBJECT (alsa, "period time %u", period_time);

   428   }

   429 

   430   /* Set buffer size and period size manually for SPDIF */

   431   if (G_UNLIKELY (alsa->iec958)) {

   432     snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;

   433     snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;

   434 

   435     CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,

   436             &buffer_size), buffer_size);

   437     CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,

   438             &period_size, NULL), period_size);

   439   }

   440 

   441   /* write the parameters to device */

   442   CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);

   443 

   444   /* now get the configured values */

   445   CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),

   446       buffer_size);

   447   CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, &dir),

   448       period_size);

   449 

   450   GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,

   451       alsa->period_size);

   452 

   453   snd_pcm_hw_params_free (params);

   454   return 0;

   455 

   456   /* ERRORS */

   457 no_config:

   458   {

   459     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   460         ("Broken configuration for playback: no configurations available: %s",

   461             snd_strerror (err)));

   462     snd_pcm_hw_params_free (params);

   463     return err;

   464   }

   465 wrong_access:

   466   {

   467     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   468         ("Access type not available for playback: %s", snd_strerror (err)));

   469     snd_pcm_hw_params_free (params);

   470     return err;

   471   }

   472 no_sample_format:

   473   {

   474     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   475         ("Sample format not available for playback: %s", snd_strerror (err)));

   476     snd_pcm_hw_params_free (params);

   477     return err;

   478   }

   479 no_channels:

   480   {

   481     gchar *msg = NULL;

   482 

   483     if ((alsa->channels) == 1)

   484       msg = g_strdup (_("Could not open device for playback in mono mode."));

   485     if ((alsa->channels) == 2)

   486       msg = g_strdup (_("Could not open device for playback in stereo mode."));

   487     if ((alsa->channels) > 2)

   488       msg =

   489           g_strdup_printf (_

   490           ("Could not open device for playback in %d-channel mode."),

   491           alsa->channels);

   492     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (msg), (snd_strerror (err)));

   493     g_free (msg);

   494     snd_pcm_hw_params_free (params);

   495     return err;

   496   }

   497 no_rate:

   498   {

   499     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   500         ("Rate %iHz not available for playback: %s",

   501             alsa->rate, snd_strerror (err)));

   502     return err;

   503   }

   504 rate_match:

   505   {

   506     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   507         ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));

   508     snd_pcm_hw_params_free (params);

   509     return -EINVAL;

   510   }

   511 buffer_size:

   512   {

   513     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   514         ("Unable to get buffer size for playback: %s", snd_strerror (err)));

   515     snd_pcm_hw_params_free (params);

   516     return err;

   517   }

   518 period_size:

   519   {

   520     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   521         ("Unable to get period size for playback: %s", snd_strerror (err)));

   522     snd_pcm_hw_params_free (params);

   523     return err;

   524   }

   525 set_hw_params:

   526   {

   527     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   528         ("Unable to set hw params for playback: %s", snd_strerror (err)));

   529     snd_pcm_hw_params_free (params);

   530     return err;

   531   }

   532 }

   533 

   534 static int

   535 set_swparams (GstAlsaSink * alsa)

   536 {

   537   int err;

   538   snd_pcm_sw_params_t *params;

   539 

   540   snd_pcm_sw_params_malloc (&params);

   541 

   542   /* get the current swparams */

   543   CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);

   544   /* start the transfer when the buffer is almost full: */

   545   /* (buffer_size / avail_min) * avail_min */

   546   CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,

   547           (alsa->buffer_size / alsa->period_size) * alsa->period_size),

   548       start_threshold);

   549 

   550   /* allow the transfer when at least period_size samples can be processed */

   551   CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,

   552           alsa->period_size), set_avail);

   553 

   554 #if GST_CHECK_ALSA_VERSION(1,0,16)

   555   /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */

   556 #else

   557   /* align all transfers to 1 sample */

   558   CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);

   559 #endif

   560 

   561   /* write the parameters to the playback device */

   562   CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);

   563 

   564   snd_pcm_sw_params_free (params);

   565   return 0;

   566 

   567   /* ERRORS */

   568 no_config:

   569   {

   570     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   571         ("Unable to determine current swparams for playback: %s",

   572             snd_strerror (err)));

   573     snd_pcm_sw_params_free (params);

   574     return err;

   575   }

   576 start_threshold:

   577   {

   578     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   579         ("Unable to set start threshold mode for playback: %s",

   580             snd_strerror (err)));

   581     snd_pcm_sw_params_free (params);

   582     return err;

   583   }

   584 set_avail:

   585   {

   586     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   587         ("Unable to set avail min for playback: %s", snd_strerror (err)));

   588     snd_pcm_sw_params_free (params);

   589     return err;

   590   }

   591 #if !GST_CHECK_ALSA_VERSION(1,0,16)

   592 set_align:

   593   {

   594     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   595         ("Unable to set transfer align for playback: %s", snd_strerror (err)));

   596     snd_pcm_sw_params_free (params);

   597     return err;

   598   }

   599 #endif

   600 set_sw_params:

   601   {

   602     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   603         ("Unable to set sw params for playback: %s", snd_strerror (err)));

   604     snd_pcm_sw_params_free (params);

   605     return err;

   606   }

   607 }

   608 

   609 static gboolean

   610 alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec)

   611 {

   612   /* Initialize our boolean */

   613   alsa->iec958 = FALSE;

   614 

   615   switch (spec->type) {

   616     case GST_BUFTYPE_LINEAR:

   617       GST_DEBUG_OBJECT (alsa,

   618           "Linear format : depth=%d, width=%d, sign=%d, bigend=%d", spec->depth,

   619           spec->width, spec->sign, spec->bigend);

   620 

   621       alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,

   622           spec->sign ? 0 : 1, spec->bigend ? 1 : 0);

   623       break;

   624     case GST_BUFTYPE_FLOAT:

   625       switch (spec->format) {

   626         case GST_FLOAT32_LE:

   627           alsa->format = SND_PCM_FORMAT_FLOAT_LE;

   628           break;

   629         case GST_FLOAT32_BE:

   630           alsa->format = SND_PCM_FORMAT_FLOAT_BE;

   631           break;

   632         case GST_FLOAT64_LE:

   633           alsa->format = SND_PCM_FORMAT_FLOAT64_LE;

   634           break;

   635         case GST_FLOAT64_BE:

   636           alsa->format = SND_PCM_FORMAT_FLOAT64_BE;

   637           break;

   638         default:

   639           goto error;

   640       }

   641       break;

   642     case GST_BUFTYPE_A_LAW:

   643       alsa->format = SND_PCM_FORMAT_A_LAW;

   644       break;

   645     case GST_BUFTYPE_MU_LAW:

   646       alsa->format = SND_PCM_FORMAT_MU_LAW;

   647       break;

   648     case GST_BUFTYPE_IEC958:

   649       alsa->format = SND_PCM_FORMAT_S16_BE;

   650       alsa->iec958 = TRUE;

   651       break;

   652     default:

   653       goto error;

   654 

   655   }

   656   alsa->rate = spec->rate;

   657   alsa->channels = spec->channels;

   658   alsa->buffer_time = spec->buffer_time;

   659   alsa->period_time = spec->latency_time;

   660   alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;

   661 

   662   return TRUE;

   663 

   664   /* ERRORS */

   665 error:

   666   {

   667     return FALSE;

   668   }

   669 }

   670 

   671 static gboolean

   672 gst_alsasink_open (GstAudioSink * asink)

   673 {

   674   GstAlsaSink *alsa;

   675   gint err;

   676 

   677   alsa = GST_ALSA_SINK (asink);

   678 

   679   CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,

   680           SND_PCM_NONBLOCK), open_error);

   681   GST_LOG_OBJECT (alsa, "Opened device %s", alsa->device);

   682 

   683   return TRUE;

   684 

   685   /* ERRORS */

   686 open_error:

   687   {

   688     if (err == -EBUSY) {

   689       GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,

   690           (_("Could not open audio device for playback. "

   691                   "Device is being used by another application.")),

   692           ("Device '%s' is busy", alsa->device));

   693     } else {

   694       GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,

   695           (_("Could not open audio device for playback.")),

   696           ("Playback open error on device '%s': %s", alsa->device,

   697               snd_strerror (err)));

   698     }

   699     return FALSE;

   700   }

   701 }

   702 

   703 static gboolean

   704 gst_alsasink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)

   705 {

   706   GstAlsaSink *alsa;

   707   gint err;

   708 

   709   alsa = GST_ALSA_SINK (asink);

   710 

   711   if (spec->format == GST_IEC958) {

   712     snd_pcm_close (alsa->handle);

   713     alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa));

   714     if (G_UNLIKELY (!alsa->handle)) {

   715       goto no_iec958;

   716     }

   717   }

   718 

   719   if (!alsasink_parse_spec (alsa, spec))

   720     goto spec_parse;

   721 

   722   CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);

   723 

   724   CHECK (set_hwparams (alsa), hw_params_failed);

   725   CHECK (set_swparams (alsa), sw_params_failed);

   726 

   727   alsa->bytes_per_sample = spec->bytes_per_sample;

   728   spec->segsize = alsa->period_size * spec->bytes_per_sample;

   729   spec->segtotal = alsa->buffer_size / alsa->period_size;

   730 

   731   {

   732     snd_output_t *out_buf = NULL;

   733     char *msg = NULL;

   734 

   735     snd_output_buffer_open (&out_buf);

   736     snd_pcm_dump_hw_setup (alsa->handle, out_buf);

   737     snd_output_buffer_string (out_buf, &msg);

   738     GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);

   739     snd_output_close (out_buf);

   740     snd_output_buffer_open (&out_buf);

   741     snd_pcm_dump_sw_setup (alsa->handle, out_buf);

   742     snd_output_buffer_string (out_buf, &msg);

   743     GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);

   744     snd_output_close (out_buf);

   745   }

   746 

   747   return TRUE;

   748 

   749   /* ERRORS */

   750 no_iec958:

   751   {

   752     GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL),

   753         ("Could not open IEC958 (SPDIF) device for playback"));

   754     return FALSE;

   755   }

   756 spec_parse:

   757   {

   758     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   759         ("Error parsing spec"));

   760     return FALSE;

   761   }

   762 non_block:

   763   {

   764     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   765         ("Could not set device to blocking: %s", snd_strerror (err)));

   766     return FALSE;

   767   }

   768 hw_params_failed:

   769   {

   770     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   771         ("Setting of hwparams failed: %s", snd_strerror (err)));

   772     return FALSE;

   773   }

   774 sw_params_failed:

   775   {

   776     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   777         ("Setting of swparams failed: %s", snd_strerror (err)));

   778     return FALSE;

   779   }

   780 }

   781 

   782 static gboolean

   783 gst_alsasink_unprepare (GstAudioSink * asink)

   784 {

   785   GstAlsaSink *alsa;

   786   gint err;

   787 

   788   alsa = GST_ALSA_SINK (asink);

   789 

   790   CHECK (snd_pcm_drop (alsa->handle), drop);

   791 

   792   CHECK (snd_pcm_hw_free (alsa->handle), hw_free);

   793 

   794   CHECK (snd_pcm_nonblock (alsa->handle, 1), non_block);

   795 

   796   return TRUE;

   797 

   798   /* ERRORS */

   799 drop:

   800   {

   801     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   802         ("Could not drop samples: %s", snd_strerror (err)));

   803     return FALSE;

   804   }

   805 hw_free:

   806   {

   807     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   808         ("Could not free hw params: %s", snd_strerror (err)));

   809     return FALSE;

   810   }

   811 non_block:

   812   {

   813     GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

   814         ("Could not set device to nonblocking: %s", snd_strerror (err)));

   815     return FALSE;

   816   }

   817 }

   818 

   819 static gboolean

   820 gst_alsasink_close (GstAudioSink * asink)

   821 {

   822   GstAlsaSink *alsa = GST_ALSA_SINK (asink);

   823   gint err;

   824 

   825   if (alsa->handle) {

   826     CHECK (snd_pcm_close (alsa->handle), close_error);

   827     alsa->handle = NULL;

   828   }

   829   gst_caps_replace (&alsa->cached_caps, NULL);

   830 

   831   return TRUE;

   832 

   833   /* ERRORS */

   834 close_error:

   835   {

   836     GST_ELEMENT_ERROR (alsa, RESOURCE, CLOSE, (NULL),

   837         ("Playback close error: %s", snd_strerror (err)));

   838     return FALSE;

   839   }

   840 }

   841 

   842 

   843 /*

   844  *   Underrun and suspend recovery

   845  */

   846 static gint

   847 xrun_recovery (GstAlsaSink * alsa, snd_pcm_t * handle, gint err)

   848 {

   849   GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);

   850 

   851   if (err == -EPIPE) {          /* under-run */

   852     err = snd_pcm_prepare (handle);

   853     if (err < 0)

   854       GST_WARNING_OBJECT (alsa,

   855           "Can't recovery from underrun, prepare failed: %s",

   856           snd_strerror (err));

   857     return 0;

   858   } else if (err == -ESTRPIPE) {

   859     while ((err = snd_pcm_resume (handle)) == -EAGAIN)

   860       g_usleep (100);           /* wait until the suspend flag is released */

   861 

   862     if (err < 0) {

   863       err = snd_pcm_prepare (handle);

   864       if (err < 0)

   865         GST_WARNING_OBJECT (alsa,

   866             "Can't recovery from suspend, prepare failed: %s",

   867             snd_strerror (err));

   868     }

   869     return 0;

   870   }

   871   return err;

   872 }

   873 

   874 static guint

   875 gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)

   876 {

   877   GstAlsaSink *alsa;

   878   gint err;

   879   gint cptr;

   880   gint16 *ptr = data;

   881 

   882   alsa = GST_ALSA_SINK (asink);

   883 

   884   if (alsa->iec958 && alsa->need_swap) {

   885     guint i;

   886 

   887     GST_DEBUG_OBJECT (asink, "swapping bytes");

   888     for (i = 0; i < length / 2; i++) {

   889       ptr[i] = GUINT16_SWAP_LE_BE (ptr[i]);

   890     }

   891   }

   892 

   893   GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length);

   894 

   895   cptr = length / alsa->bytes_per_sample;

   896 

   897   GST_ALSA_SINK_LOCK (asink);

   898   while (cptr > 0) {

   899     err = snd_pcm_writei (alsa->handle, ptr, cptr);

   900 

   901     GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr);

   902     if (err < 0) {

   903       GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err));

   904       if (err == -EAGAIN) {

   905         continue;

   906       } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {

   907         goto write_error;

   908       }

   909       continue;

   910     }

   911 

   912     ptr += snd_pcm_frames_to_bytes (alsa->handle, err);

   913     cptr -= err;

   914   }

   915   GST_ALSA_SINK_UNLOCK (asink);

   916 

   917   return length - (cptr * alsa->bytes_per_sample);

   918 

   919 write_error:

   920   {

   921     GST_ALSA_SINK_UNLOCK (asink);

   922     return length;              /* skip one period */

   923   }

   924 }

   925 

   926 static guint

   927 gst_alsasink_delay (GstAudioSink * asink)

   928 {

   929   GstAlsaSink *alsa;

   930   snd_pcm_sframes_t delay;

   931   int res;

   932 

   933   alsa = GST_ALSA_SINK (asink);

   934 

   935   res = snd_pcm_delay (alsa->handle, &delay);

   936   if (G_UNLIKELY (res < 0)) {

   937     /* on errors, report 0 delay */

   938     GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);

   939     delay = 0;

   940   }

   941   if (G_UNLIKELY (delay < 0)) {

   942     /* make sure we never return a negative delay */

   943     GST_WARNING_OBJECT (alsa, "snd_pcm_delay returned negative delay");

   944     delay = 0;

   945   }

   946 

   947   return delay;

   948 }

   949 

   950 static void

   951 gst_alsasink_reset (GstAudioSink * asink)

   952 {

   953   GstAlsaSink *alsa;

   954   gint err;

   955 

   956   alsa = GST_ALSA_SINK (asink);

   957 

   958   GST_ALSA_SINK_LOCK (asink);

   959   GST_DEBUG_OBJECT (alsa, "drop");

   960   CHECK (snd_pcm_drop (alsa->handle), drop_error);

   961   GST_DEBUG_OBJECT (alsa, "prepare");

   962   CHECK (snd_pcm_prepare (alsa->handle), prepare_error);

   963   GST_DEBUG_OBJECT (alsa, "reset done");

   964   GST_ALSA_SINK_UNLOCK (asink);

   965 

   966   return;

   967 

   968   /* ERRORS */

   969 drop_error:

   970   {

   971     GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",

   972         snd_strerror (err));

   973     GST_ALSA_SINK_UNLOCK (asink);

   974     return;

   975   }

   976 prepare_error:

   977   {

   978     GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",

   979         snd_strerror (err));

   980     GST_ALSA_SINK_UNLOCK (asink);

   981     return;

   982   }

   983 }