--- /dev/null
+/* GStreamer
+ * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
+ * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
+ *
+ * gstalsasink.c:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-alsasink
+ * @see_also: alsasrc, alsamixer
+ *
+ * This element renders raw audio samples using the ALSA api.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink
+ * ]| Play an Ogg/Vorbis file.
+ * </refsect2>
+ *
+ * Last reviewed on 2006-03-01 (0.10.4)
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#include <string.h>
+#include <getopt.h>
+#include <alsa/asoundlib.h>
+
+#include "gstalsa.h"
+#include "gstalsasink.h"
+#include "gstalsadeviceprobe.h"
+
+#include <gst/gst-i18n-plugin.h>
+
+#define DEFAULT_DEVICE "default"
+#define DEFAULT_DEVICE_NAME ""
+#define DEFAULT_CARD_NAME ""
+#define SPDIF_PERIOD_SIZE 1536
+#define SPDIF_BUFFER_SIZE 15360
+
+enum
+{
+ PROP_0,
+ PROP_DEVICE,
+ PROP_DEVICE_NAME,
+ PROP_CARD_NAME,
+ PROP_LAST
+};
+
+static void gst_alsasink_init_interfaces (GType type);
+
+GST_BOILERPLATE_FULL (GstAlsaSink, gst_alsasink, GstAudioSink,
+ GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces);
+
+static void gst_alsasink_finalise (GObject * object);
+static void gst_alsasink_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_alsasink_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink);
+
+static gboolean gst_alsasink_open (GstAudioSink * asink);
+static gboolean gst_alsasink_prepare (GstAudioSink * asink,
+ GstRingBufferSpec * spec);
+static gboolean gst_alsasink_unprepare (GstAudioSink * asink);
+static gboolean gst_alsasink_close (GstAudioSink * asink);
+static guint gst_alsasink_write (GstAudioSink * asink, gpointer data,
+ guint length);
+static guint gst_alsasink_delay (GstAudioSink * asink);
+static void gst_alsasink_reset (GstAudioSink * asink);
+
+static gint output_ref; /* 0 */
+static snd_output_t *output; /* NULL */
+static GStaticMutex output_mutex = G_STATIC_MUTEX_INIT;
+
+
+#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
+# define ALSA_SINK_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
+#else
+# define ALSA_SINK_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
+#endif
+
+static GstStaticPadTemplate alsasink_sink_factory =
+ GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
+ "signed = (boolean) { TRUE, FALSE }, "
+ "width = (int) 32, "
+ "depth = (int) 32, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
+ "audio/x-raw-int, "
+ "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
+ "signed = (boolean) { TRUE, FALSE }, "
+ "width = (int) 24, "
+ "depth = (int) 24, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
+ "audio/x-raw-int, "
+ "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
+ "signed = (boolean) { TRUE, FALSE }, "
+ "width = (int) 32, "
+ "depth = (int) 24, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
+ "audio/x-raw-int, "
+ "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "
+ "signed = (boolean) { TRUE, FALSE }, "
+ "width = (int) 16, "
+ "depth = (int) 16, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; "
+ "audio/x-raw-int, "
+ "signed = (boolean) { TRUE, FALSE }, "
+ "width = (int) 8, "
+ "depth = (int) 8, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ];"
+ "audio/x-iec958")
+ );
+
+static void
+gst_alsasink_finalise (GObject * object)
+{
+ GstAlsaSink *sink = GST_ALSA_SINK (object);
+
+ g_free (sink->device);
+ g_mutex_free (sink->alsa_lock);
+
+ g_static_mutex_lock (&output_mutex);
+ --output_ref;
+ if (output_ref == 0) {
+ snd_output_close (output);
+ output = NULL;
+ }
+ g_static_mutex_unlock (&output_mutex);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_alsasink_init_interfaces (GType type)
+{
+ gst_alsa_type_add_device_property_probe_interface (type);
+}
+
+static void
+gst_alsasink_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+ gst_element_class_set_details_simple (element_class,
+ "Audio sink (ALSA)", "Sink/Audio",
+ "Output to a sound card via ALSA", "Wim Taymans <wim@fluendo.com>");
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&alsasink_sink_factory));
+}
+
+static void
+gst_alsasink_class_init (GstAlsaSinkClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstBaseSinkClass *gstbasesink_class;
+ GstAudioSinkClass *gstaudiosink_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstbasesink_class = (GstBaseSinkClass *) klass;
+ gstaudiosink_class = (GstAudioSinkClass *) klass;
+
+ parent_class = g_type_class_peek_parent (klass);
+
+ gobject_class->finalize = gst_alsasink_finalise;
+ gobject_class->get_property = gst_alsasink_get_property;
+ gobject_class->set_property = gst_alsasink_set_property;
+
+ gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);
+
+ gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open);
+ gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare);
+ gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare);
+ gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close);
+ gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write);
+ gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay);
+ gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink_reset);
+
+ g_object_class_install_property (gobject_class, PROP_DEVICE,
+ g_param_spec_string ("device", "Device",
+ "ALSA device, as defined in an asound configuration file",
+ DEFAULT_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
+ g_param_spec_string ("device-name", "Device name",
+ "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_CARD_NAME,
+ g_param_spec_string ("card-name", "Card name",
+ "Human-readable name of the sound card", DEFAULT_CARD_NAME,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_alsasink_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstAlsaSink *sink;
+
+ sink = GST_ALSA_SINK (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ g_free (sink->device);
+ sink->device = g_value_dup_string (value);
+ /* setting NULL restores the default device */
+ if (sink->device == NULL) {
+ sink->device = g_strdup (DEFAULT_DEVICE);
+ }
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_alsasink_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstAlsaSink *sink;
+
+ sink = GST_ALSA_SINK (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ g_value_set_string (value, sink->device);
+ break;
+ case PROP_DEVICE_NAME:
+ g_value_take_string (value,
+ gst_alsa_find_device_name (GST_OBJECT_CAST (sink),
+ sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK));
+ break;
+ case PROP_CARD_NAME:
+ g_value_take_string (value,
+ gst_alsa_find_card_name (GST_OBJECT_CAST (sink),
+ sink->device, SND_PCM_STREAM_PLAYBACK));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_alsasink_init (GstAlsaSink * alsasink, GstAlsaSinkClass * g_class)
+{
+ GST_DEBUG_OBJECT (alsasink, "initializing alsasink");
+
+ alsasink->device = g_strdup (DEFAULT_DEVICE);
+ alsasink->handle = NULL;
+ alsasink->cached_caps = NULL;
+ alsasink->alsa_lock = g_mutex_new ();
+
+ g_static_mutex_lock (&output_mutex);
+ if (output_ref == 0) {
+ snd_output_stdio_attach (&output, stdout, 0);
+ ++output_ref;
+ }
+ g_static_mutex_unlock (&output_mutex);
+}
+
+#define CHECK(call, error) \
+G_STMT_START { \
+if ((err = call) < 0) \
+ goto error; \
+} G_STMT_END;
+
+static GstCaps *
+gst_alsasink_getcaps (GstBaseSink * bsink)
+{
+ GstElementClass *element_class;
+ GstPadTemplate *pad_template;
+ GstAlsaSink *sink = GST_ALSA_SINK (bsink);
+ GstCaps *caps;
+
+ if (sink->handle == NULL) {
+ GST_DEBUG_OBJECT (sink, "device not open, using template caps");
+ return NULL; /* base class will get template caps for us */
+ }
+
+ if (sink->cached_caps) {
+ GST_LOG_OBJECT (sink, "Returning cached caps");
+ return gst_caps_ref (sink->cached_caps);
+ }
+
+ element_class = GST_ELEMENT_GET_CLASS (sink);
+ pad_template = gst_element_class_get_pad_template (element_class, "sink");
+ g_return_val_if_fail (pad_template != NULL, NULL);
+
+ caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->handle,
+ gst_pad_template_get_caps (pad_template));
+
+ if (caps) {
+ sink->cached_caps = gst_caps_ref (caps);
+ }
+
+ GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);
+
+ return caps;
+}
+
+static int
+set_hwparams (GstAlsaSink * alsa)
+{
+ guint rrate;
+ gint err;
+ snd_pcm_hw_params_t *params;
+ guint period_time, buffer_time;
+
+ snd_pcm_hw_params_malloc (¶ms);
+
+ GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "
+ "SPDIF (%d)", alsa->channels, alsa->rate,
+ snd_pcm_format_name (alsa->format), alsa->iec958);
+
+ /* start with requested values, if we cannot configure alsa for those values,
+ * we set these values to -1, which will leave the default alsa values */
+ buffer_time = alsa->buffer_time;
+ period_time = alsa->period_time;
+
+retry:
+ /* choose all parameters */
+ CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
+ /* set the interleaved read/write format */
+ CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
+ wrong_access);
+ /* set the sample format */
+ if (alsa->iec958) {
+ /* Try to use big endian first else fallback to le and swap bytes */
+ if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {
+ alsa->format = SND_PCM_FORMAT_S16_LE;
+ alsa->need_swap = TRUE;
+ GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");
+ } else {
+ alsa->need_swap = FALSE;
+ }
+ }
+ CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
+ no_sample_format);
+ /* set the count of channels */
+ CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
+ no_channels);
+ /* set the stream rate */
+ rrate = alsa->rate;
+ CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
+ no_rate);
+ if (rrate != alsa->rate)
+ goto rate_match;
+
+#ifndef GST_DISABLE_GST_DEBUG
+ /* get and dump some limits */
+ {
+ guint min, max;
+
+ snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL);
+ snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);
+
+ GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",
+ alsa->buffer_time, min, max);
+
+ snd_pcm_hw_params_get_period_time_min (params, &min, NULL);
+ snd_pcm_hw_params_get_period_time_max (params, &max, NULL);
+
+ GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",
+ alsa->period_time, min, max);
+
+ snd_pcm_hw_params_get_periods_min (params, &min, NULL);
+ snd_pcm_hw_params_get_periods_max (params, &max, NULL);
+
+ GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
+ }
+#endif
+
+ /* now try to configure the buffer time and period time, if one
+ * of those fail, we fall back to the defaults and emit a warning. */
+ if (buffer_time != -1 && !alsa->iec958) {
+ /* set the buffer time */
+ if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
+ &buffer_time, NULL)) < 0) {
+ GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to set buffer time %i for playback: %s",
+ buffer_time, snd_strerror (err)));
+ /* disable buffer_time the next round */
+ buffer_time = -1;
+ goto retry;
+ }
+ GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time);
+ }
+ if (period_time != -1 && !alsa->iec958) {
+ /* set the period time */
+ if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
+ &period_time, NULL)) < 0) {
+ GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to set period time %i for playback: %s",
+ period_time, snd_strerror (err)));
+ /* disable period_time the next round */
+ period_time = -1;
+ goto retry;
+ }
+ GST_DEBUG_OBJECT (alsa, "period time %u", period_time);
+ }
+
+ /* Set buffer size and period size manually for SPDIF */
+ if (G_UNLIKELY (alsa->iec958)) {
+ snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;
+ snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;
+
+ CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,
+ &buffer_size), buffer_size);
+ CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,
+ &period_size, NULL), period_size);
+ }
+
+ /* write the parameters to device */
+ CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
+
+ /* now get the configured values */
+ CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
+ buffer_size);
+ CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
+ period_size);
+
+ GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
+ alsa->period_size);
+
+ snd_pcm_hw_params_free (params);
+ return 0;
+
+ /* ERRORS */
+no_config:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Broken configuration for playback: no configurations available: %s",
+ snd_strerror (err)));
+ snd_pcm_hw_params_free (params);
+ return err;
+ }
+wrong_access:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Access type not available for playback: %s", snd_strerror (err)));
+ snd_pcm_hw_params_free (params);
+ return err;
+ }
+no_sample_format:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Sample format not available for playback: %s", snd_strerror (err)));
+ snd_pcm_hw_params_free (params);
+ return err;
+ }
+no_channels:
+ {
+ gchar *msg = NULL;
+
+ if ((alsa->channels) == 1)
+ msg = g_strdup (_("Could not open device for playback in mono mode."));
+ if ((alsa->channels) == 2)
+ msg = g_strdup (_("Could not open device for playback in stereo mode."));
+ if ((alsa->channels) > 2)
+ msg =
+ g_strdup_printf (_
+ ("Could not open device for playback in %d-channel mode."),
+ alsa->channels);
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
+ ("%s", snd_strerror (err)));
+ g_free (msg);
+ snd_pcm_hw_params_free (params);
+ return err;
+ }
+no_rate:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Rate %iHz not available for playback: %s",
+ alsa->rate, snd_strerror (err)));
+ return err;
+ }
+rate_match:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));
+ snd_pcm_hw_params_free (params);
+ return -EINVAL;
+ }
+buffer_size:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to get buffer size for playback: %s", snd_strerror (err)));
+ snd_pcm_hw_params_free (params);
+ return err;
+ }
+period_size:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to get period size for playback: %s", snd_strerror (err)));
+ snd_pcm_hw_params_free (params);
+ return err;
+ }
+set_hw_params:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to set hw params for playback: %s", snd_strerror (err)));
+ snd_pcm_hw_params_free (params);
+ return err;
+ }
+}
+
+static int
+set_swparams (GstAlsaSink * alsa)
+{
+ int err;
+ snd_pcm_sw_params_t *params;
+
+ snd_pcm_sw_params_malloc (¶ms);
+
+ /* get the current swparams */
+ CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
+ /* start the transfer when the buffer is almost full: */
+ /* (buffer_size / avail_min) * avail_min */
+ CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
+ (alsa->buffer_size / alsa->period_size) * alsa->period_size),
+ start_threshold);
+
+ /* allow the transfer when at least period_size samples can be processed */
+ CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
+ alsa->period_size), set_avail);
+
+#if GST_CHECK_ALSA_VERSION(1,0,16)
+ /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
+#else
+ /* align all transfers to 1 sample */
+ CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
+#endif
+
+ /* write the parameters to the playback device */
+ CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);
+
+ snd_pcm_sw_params_free (params);
+ return 0;
+
+ /* ERRORS */
+no_config:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to determine current swparams for playback: %s",
+ snd_strerror (err)));
+ snd_pcm_sw_params_free (params);
+ return err;
+ }
+start_threshold:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to set start threshold mode for playback: %s",
+ snd_strerror (err)));
+ snd_pcm_sw_params_free (params);
+ return err;
+ }
+set_avail:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to set avail min for playback: %s", snd_strerror (err)));
+ snd_pcm_sw_params_free (params);
+ return err;
+ }
+#if !GST_CHECK_ALSA_VERSION(1,0,16)
+set_align:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to set transfer align for playback: %s", snd_strerror (err)));
+ snd_pcm_sw_params_free (params);
+ return err;
+ }
+#endif
+set_sw_params:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Unable to set sw params for playback: %s", snd_strerror (err)));
+ snd_pcm_sw_params_free (params);
+ return err;
+ }
+}
+
+static gboolean
+alsasink_parse_spec (GstAlsaSink * alsa, GstRingBufferSpec * spec)
+{
+ /* Initialize our boolean */
+ alsa->iec958 = FALSE;
+
+ switch (spec->type) {
+ case GST_BUFTYPE_LINEAR:
+ GST_DEBUG_OBJECT (alsa,
+ "Linear format : depth=%d, width=%d, sign=%d, bigend=%d", spec->depth,
+ spec->width, spec->sign, spec->bigend);
+
+ alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width,
+ spec->sign ? 0 : 1, spec->bigend ? 1 : 0);
+ break;
+ case GST_BUFTYPE_FLOAT:
+ switch (spec->format) {
+ case GST_FLOAT32_LE:
+ alsa->format = SND_PCM_FORMAT_FLOAT_LE;
+ break;
+ case GST_FLOAT32_BE:
+ alsa->format = SND_PCM_FORMAT_FLOAT_BE;
+ break;
+ case GST_FLOAT64_LE:
+ alsa->format = SND_PCM_FORMAT_FLOAT64_LE;
+ break;
+ case GST_FLOAT64_BE:
+ alsa->format = SND_PCM_FORMAT_FLOAT64_BE;
+ break;
+ default:
+ goto error;
+ }
+ break;
+ case GST_BUFTYPE_A_LAW:
+ alsa->format = SND_PCM_FORMAT_A_LAW;
+ break;
+ case GST_BUFTYPE_MU_LAW:
+ alsa->format = SND_PCM_FORMAT_MU_LAW;
+ break;
+ case GST_BUFTYPE_IEC958:
+ alsa->format = SND_PCM_FORMAT_S16_BE;
+ alsa->iec958 = TRUE;
+ break;
+ default:
+ goto error;
+
+ }
+ alsa->rate = spec->rate;
+ alsa->channels = spec->channels;
+ alsa->buffer_time = spec->buffer_time;
+ alsa->period_time = spec->latency_time;
+ alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED;
+
+ return TRUE;
+
+ /* ERRORS */
+error:
+ {
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_alsasink_open (GstAudioSink * asink)
+{
+ GstAlsaSink *alsa;
+ gint err;
+
+ alsa = GST_ALSA_SINK (asink);
+
+ /* open in non-blocking mode, we'll use snd_pcm_wait() for space to become
+ * available. */
+ CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK), open_error);
+ GST_LOG_OBJECT (alsa, "Opened device %s", alsa->device);
+
+ return TRUE;
+
+ /* ERRORS */
+open_error:
+ {
+ if (err == -EBUSY) {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY,
+ (_("Could not open audio device for playback. "
+ "Device is being used by another application.")),
+ ("Device '%s' is busy", alsa->device));
+ } else {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE,
+ (_("Could not open audio device for playback.")),
+ ("Playback open error on device '%s': %s", alsa->device,
+ snd_strerror (err)));
+ }
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_alsasink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
+{
+ GstAlsaSink *alsa;
+ gint err;
+
+ alsa = GST_ALSA_SINK (asink);
+
+ if (spec->format == GST_IEC958) {
+ snd_pcm_close (alsa->handle);
+ alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa));
+ if (G_UNLIKELY (!alsa->handle)) {
+ goto no_iec958;
+ }
+ }
+
+ if (!alsasink_parse_spec (alsa, spec))
+ goto spec_parse;
+
+ CHECK (set_hwparams (alsa), hw_params_failed);
+ CHECK (set_swparams (alsa), sw_params_failed);
+
+ alsa->bytes_per_sample = spec->bytes_per_sample;
+ spec->segsize = alsa->period_size * spec->bytes_per_sample;
+ spec->segtotal = alsa->buffer_size / alsa->period_size;
+
+ {
+ snd_output_t *out_buf = NULL;
+ char *msg = NULL;
+
+ snd_output_buffer_open (&out_buf);
+ snd_pcm_dump_hw_setup (alsa->handle, out_buf);
+ snd_output_buffer_string (out_buf, &msg);
+ GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
+ snd_output_close (out_buf);
+ snd_output_buffer_open (&out_buf);
+ snd_pcm_dump_sw_setup (alsa->handle, out_buf);
+ snd_output_buffer_string (out_buf, &msg);
+ GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
+ snd_output_close (out_buf);
+ }
+
+ return TRUE;
+
+ /* ERRORS */
+no_iec958:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL),
+ ("Could not open IEC958 (SPDIF) device for playback"));
+ return FALSE;
+ }
+spec_parse:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Error parsing spec"));
+ return FALSE;
+ }
+hw_params_failed:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Setting of hwparams failed: %s", snd_strerror (err)));
+ return FALSE;
+ }
+sw_params_failed:
+ {
+ GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
+ ("Setting of swparams failed: %s", snd_strerror (err)));
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_alsasink_unprepare (GstAudioSink * asink)
+{
+ GstAlsaSink *alsa;
+
+ alsa = GST_ALSA_SINK (asink);
+
+ snd_pcm_drop (alsa->handle);
+ snd_pcm_hw_free (alsa->handle);
+
+ return TRUE;
+}
+
+static gboolean
+gst_alsasink_close (GstAudioSink * asink)
+{
+ GstAlsaSink *alsa = GST_ALSA_SINK (asink);
+
+ if (alsa->handle) {
+ snd_pcm_close (alsa->handle);
+ alsa->handle = NULL;
+ }
+ gst_caps_replace (&alsa->cached_caps, NULL);
+
+ return TRUE;
+}
+
+
+/*
+ * Underrun and suspend recovery
+ */
+static gint
+xrun_recovery (GstAlsaSink * alsa, snd_pcm_t * handle, gint err)
+{
+ GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err);
+
+ if (err == -EPIPE) { /* under-run */
+ err = snd_pcm_prepare (handle);
+ if (err < 0)
+ GST_WARNING_OBJECT (alsa,
+ "Can't recovery from underrun, prepare failed: %s",
+ snd_strerror (err));
+ return 0;
+ } else if (err == -ESTRPIPE) {
+ while ((err = snd_pcm_resume (handle)) == -EAGAIN)
+ g_usleep (100); /* wait until the suspend flag is released */
+
+ if (err < 0) {
+ err = snd_pcm_prepare (handle);
+ if (err < 0)
+ GST_WARNING_OBJECT (alsa,
+ "Can't recovery from suspend, prepare failed: %s",
+ snd_strerror (err));
+ }
+ return 0;
+ }
+ return err;
+}
+
+static guint
+gst_alsasink_write (GstAudioSink * asink, gpointer data, guint length)
+{
+ GstAlsaSink *alsa;
+ gint err;
+ gint cptr;
+ gint16 *ptr = data;
+
+ alsa = GST_ALSA_SINK (asink);
+
+ if (alsa->iec958 && alsa->need_swap) {
+ guint i;
+
+ GST_DEBUG_OBJECT (asink, "swapping bytes");
+ for (i = 0; i < length / 2; i++) {
+ ptr[i] = GUINT16_SWAP_LE_BE (ptr[i]);
+ }
+ }
+
+ GST_LOG_OBJECT (asink, "received audio samples buffer of %u bytes", length);
+
+ cptr = length / alsa->bytes_per_sample;
+
+ GST_ALSA_SINK_LOCK (asink);
+ while (cptr > 0) {
+ /* start by doing a blocking wait for free space. Set the timeout
+ * to 4 times the period time */
+ err = snd_pcm_wait (alsa->handle, (4 * alsa->period_time / 1000));
+ if (err < 0) {
+ GST_DEBUG_OBJECT (asink, "wait error, %d", err);
+ } else {
+ err = snd_pcm_writei (alsa->handle, ptr, cptr);
+ }
+
+ GST_DEBUG_OBJECT (asink, "written %d frames out of %d", err, cptr);
+ if (err < 0) {
+ GST_DEBUG_OBJECT (asink, "Write error: %s", snd_strerror (err));
+ if (err == -EAGAIN) {
+ continue;
+ } else if (xrun_recovery (alsa, alsa->handle, err) < 0) {
+ goto write_error;
+ }
+ continue;
+ }
+
+ ptr += snd_pcm_frames_to_bytes (alsa->handle, err);
+ cptr -= err;
+ }
+ GST_ALSA_SINK_UNLOCK (asink);
+
+ return length - (cptr * alsa->bytes_per_sample);
+
+write_error:
+ {
+ GST_ALSA_SINK_UNLOCK (asink);
+ return length; /* skip one period */
+ }
+}
+
+static guint
+gst_alsasink_delay (GstAudioSink * asink)
+{
+ GstAlsaSink *alsa;
+ snd_pcm_sframes_t delay;
+ int res;
+
+ alsa = GST_ALSA_SINK (asink);
+
+ res = snd_pcm_delay (alsa->handle, &delay);
+ if (G_UNLIKELY (res < 0)) {
+ /* on errors, report 0 delay */
+ GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
+ delay = 0;
+ }
+ if (G_UNLIKELY (delay < 0)) {
+ /* make sure we never return a negative delay */
+ GST_WARNING_OBJECT (alsa, "snd_pcm_delay returned negative delay");
+ delay = 0;
+ }
+
+ return delay;
+}
+
+static void
+gst_alsasink_reset (GstAudioSink * asink)
+{
+ GstAlsaSink *alsa;
+ gint err;
+
+ alsa = GST_ALSA_SINK (asink);
+
+ GST_ALSA_SINK_LOCK (asink);
+ GST_DEBUG_OBJECT (alsa, "drop");
+ CHECK (snd_pcm_drop (alsa->handle), drop_error);
+ GST_DEBUG_OBJECT (alsa, "prepare");
+ CHECK (snd_pcm_prepare (alsa->handle), prepare_error);
+ GST_DEBUG_OBJECT (alsa, "reset done");
+ GST_ALSA_SINK_UNLOCK (asink);
+
+ return;
+
+ /* ERRORS */
+drop_error:
+ {
+ GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s",
+ snd_strerror (err));
+ GST_ALSA_SINK_UNLOCK (asink);
+ return;
+ }
+prepare_error:
+ {
+ GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s",
+ snd_strerror (err));
+ GST_ALSA_SINK_UNLOCK (asink);
+ return;
+ }
+}
projects / mafwsubrenderer / blobdiff