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-rw-r--r--sys/sunaudio/gstsunaudiosink.c651
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diff --git a/sys/sunaudio/gstsunaudiosink.c b/sys/sunaudio/gstsunaudiosink.c
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+++ b/sys/sunaudio/gstsunaudiosink.c
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+/*
+ * GStreamer - SunAudio sink
+ * Copyright (C) 2004 David A. Schleef <ds@schleef.org>
+ * Copyright (C) 2005,2006 Sun Microsystems, Inc.,
+ * Brian Cameron <brian.cameron@sun.com>
+ * Copyright (C) 2006 Jan Schmidt <thaytan@mad.scientist.com>
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-sunaudiosink
+ *
+ * sunaudiosink is an audio sink designed to work with the Sun Audio
+ * interface available in Solaris.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch-1.0 audiotestsrc volume=0.5 ! sunaudiosink
+ * ]|
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <fcntl.h>
+#include <string.h>
+#include <stropts.h>
+#include <unistd.h>
+#include <sys/mman.h>
+
+#include "gstsunaudiosink.h"
+
+GST_DEBUG_CATEGORY_EXTERN (sunaudio_debug);
+#define GST_CAT_DEFAULT sunaudio_debug
+
+static void gst_sunaudiosink_base_init (gpointer g_class);
+static void gst_sunaudiosink_class_init (GstSunAudioSinkClass * klass);
+static void gst_sunaudiosink_init (GstSunAudioSink * filter);
+static void gst_sunaudiosink_dispose (GObject * object);
+static void gst_sunaudiosink_finalize (GObject * object);
+
+static void gst_sunaudiosink_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_sunaudiosink_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_sunaudiosink_getcaps (GstBaseSink * bsink);
+
+static gboolean gst_sunaudiosink_open (GstAudioSink * asink);
+static gboolean gst_sunaudiosink_close (GstAudioSink * asink);
+static gboolean gst_sunaudiosink_prepare (GstAudioSink * asink,
+ GstRingBufferSpec * spec);
+static gboolean gst_sunaudiosink_unprepare (GstAudioSink * asink);
+static guint gst_sunaudiosink_write (GstAudioSink * asink, gpointer data,
+ guint length);
+static guint gst_sunaudiosink_delay (GstAudioSink * asink);
+static void gst_sunaudiosink_reset (GstAudioSink * asink);
+
+#define DEFAULT_DEVICE "/dev/audio"
+enum
+{
+ PROP_0,
+ PROP_DEVICE,
+};
+
+static GstStaticPadTemplate gst_sunaudiosink_factory =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "endianness = (int) BYTE_ORDER, "
+ "signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16, "
+ /* [5510,48000] seems to be a Solaris limit */
+ "rate = (int) [ 5510, 48000 ], " "channels = (int) [ 1, 2 ]")
+ );
+
+static GstElementClass *parent_class = NULL;
+
+GType
+gst_sunaudiosink_get_type (void)
+{
+ static GType plugin_type = 0;
+
+ if (!plugin_type) {
+ static const GTypeInfo plugin_info = {
+ sizeof (GstSunAudioSinkClass),
+ gst_sunaudiosink_base_init,
+ NULL,
+ (GClassInitFunc) gst_sunaudiosink_class_init,
+ NULL,
+ NULL,
+ sizeof (GstSunAudioSink),
+ 0,
+ (GInstanceInitFunc) gst_sunaudiosink_init,
+ };
+
+ plugin_type = g_type_register_static (GST_TYPE_AUDIO_SINK,
+ "GstSunAudioSink", &plugin_info, 0);
+ }
+ return plugin_type;
+}
+
+static void
+gst_sunaudiosink_dispose (GObject * object)
+{
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_sunaudiosink_finalize (GObject * object)
+{
+ GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (object);
+
+ g_mutex_free (sunaudiosink->write_mutex);
+ g_cond_free (sunaudiosink->sleep_cond);
+
+ g_free (sunaudiosink->device);
+
+ if (sunaudiosink->fd != -1) {
+ close (sunaudiosink->fd);
+ sunaudiosink->fd = -1;
+ }
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_sunaudiosink_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_sunaudiosink_factory));
+ gst_element_class_set_static_metadata (element_class, "Sun Audio Sink",
+ "Sink/Audio",
+ "Audio sink for Sun Audio devices",
+ "David A. Schleef <ds@schleef.org>, "
+ "Brian Cameron <brian.cameron@sun.com>");
+}
+
+static void
+gst_sunaudiosink_class_init (GstSunAudioSinkClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseSinkClass *gstbasesink_class;
+ GstBaseAudioSinkClass *gstbaseaudiosink_class;
+ GstAudioSinkClass *gstaudiosink_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasesink_class = (GstBaseSinkClass *) klass;
+ gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
+ gstaudiosink_class = (GstAudioSinkClass *) klass;
+
+ parent_class = g_type_class_peek_parent (klass);
+
+ gobject_class->dispose = gst_sunaudiosink_dispose;
+ gobject_class->finalize = gst_sunaudiosink_finalize;
+
+ gobject_class->set_property = gst_sunaudiosink_set_property;
+ gobject_class->get_property = gst_sunaudiosink_get_property;
+
+ gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_sunaudiosink_getcaps);
+
+ gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_sunaudiosink_open);
+ gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_sunaudiosink_close);
+ gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_sunaudiosink_prepare);
+ gstaudiosink_class->unprepare =
+ GST_DEBUG_FUNCPTR (gst_sunaudiosink_unprepare);
+ gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_sunaudiosink_write);
+ gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_sunaudiosink_delay);
+ gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_sunaudiosink_reset);
+
+ g_object_class_install_property (gobject_class, PROP_DEVICE,
+ g_param_spec_string ("device", "Device", "Audio Device (/dev/audio)",
+ DEFAULT_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_sunaudiosink_init (GstSunAudioSink * sunaudiosink)
+{
+ const char *audiodev;
+
+ GST_DEBUG_OBJECT (sunaudiosink, "initializing sunaudiosink");
+
+ sunaudiosink->fd = -1;
+
+ audiodev = g_getenv ("AUDIODEV");
+ if (audiodev == NULL)
+ audiodev = DEFAULT_DEVICE;
+ sunaudiosink->device = g_strdup (audiodev);
+
+ /* mutex and gcond used to control the write method */
+ sunaudiosink->write_mutex = g_mutex_new ();
+ sunaudiosink->sleep_cond = g_cond_new ();
+}
+
+static void
+gst_sunaudiosink_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstSunAudioSink *sunaudiosink;
+
+ sunaudiosink = GST_SUNAUDIO_SINK (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ GST_OBJECT_LOCK (sunaudiosink);
+ g_free (sunaudiosink->device);
+ sunaudiosink->device = g_strdup (g_value_get_string (value));
+ GST_OBJECT_UNLOCK (sunaudiosink);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_sunaudiosink_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstSunAudioSink *sunaudiosink;
+
+ sunaudiosink = GST_SUNAUDIO_SINK (object);
+
+ switch (prop_id) {
+ case PROP_DEVICE:
+ GST_OBJECT_LOCK (sunaudiosink);
+ g_value_set_string (value, sunaudiosink->device);
+ GST_OBJECT_UNLOCK (sunaudiosink);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstCaps *
+gst_sunaudiosink_getcaps (GstBaseSink * bsink)
+{
+ GstPadTemplate *pad_template;
+ GstCaps *caps = NULL;
+ GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (bsink);
+
+ GST_DEBUG_OBJECT (sunaudiosink, "getcaps called");
+
+ pad_template = gst_static_pad_template_get (&gst_sunaudiosink_factory);
+ caps = gst_caps_copy (gst_pad_template_get_caps (pad_template));
+
+ gst_object_unref (pad_template);
+
+ return caps;
+}
+
+static gboolean
+gst_sunaudiosink_open (GstAudioSink * asink)
+{
+ GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
+ int fd, ret;
+
+ /* First try to open non-blocking */
+ GST_OBJECT_LOCK (sunaudiosink);
+ fd = open (sunaudiosink->device, O_WRONLY | O_NONBLOCK);
+
+ if (fd >= 0) {
+ close (fd);
+ fd = open (sunaudiosink->device, O_WRONLY);
+ }
+
+ if (fd == -1) {
+ GST_OBJECT_UNLOCK (sunaudiosink);
+ goto open_failed;
+ }
+
+ sunaudiosink->fd = fd;
+ GST_OBJECT_UNLOCK (sunaudiosink);
+
+ ret = ioctl (fd, AUDIO_GETDEV, &sunaudiosink->dev);
+ if (ret == -1)
+ goto ioctl_error;
+
+ GST_DEBUG_OBJECT (sunaudiosink, "name %s", sunaudiosink->dev.name);
+ GST_DEBUG_OBJECT (sunaudiosink, "version %s", sunaudiosink->dev.version);
+ GST_DEBUG_OBJECT (sunaudiosink, "config %s", sunaudiosink->dev.config);
+
+ ret = ioctl (fd, AUDIO_GETINFO, &sunaudiosink->info);
+ if (ret == -1)
+ goto ioctl_error;
+
+ GST_DEBUG_OBJECT (sunaudiosink, "monitor_gain %d",
+ sunaudiosink->info.monitor_gain);
+ GST_DEBUG_OBJECT (sunaudiosink, "output_muted %d",
+ sunaudiosink->info.output_muted);
+ GST_DEBUG_OBJECT (sunaudiosink, "hw_features %08x",
+ sunaudiosink->info.hw_features);
+ GST_DEBUG_OBJECT (sunaudiosink, "sw_features %08x",
+ sunaudiosink->info.sw_features);
+ GST_DEBUG_OBJECT (sunaudiosink, "sw_features_enabled %08x",
+ sunaudiosink->info.sw_features_enabled);
+
+ return TRUE;
+
+open_failed:
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, OPEN_WRITE, (NULL),
+ ("can't open connection to Sun Audio device %s", sunaudiosink->device));
+ return FALSE;
+ioctl_error:
+ close (sunaudiosink->fd);
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
+ strerror (errno)));
+ return FALSE;
+}
+
+static gboolean
+gst_sunaudiosink_close (GstAudioSink * asink)
+{
+ GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
+
+ if (sunaudiosink->fd != -1) {
+ close (sunaudiosink->fd);
+ sunaudiosink->fd = -1;
+ }
+ return TRUE;
+}
+
+static gboolean
+gst_sunaudiosink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
+{
+ GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
+ audio_info_t ainfo;
+ int ret;
+ int ports;
+
+ ret = ioctl (sunaudiosink->fd, AUDIO_GETINFO, &ainfo);
+ if (ret == -1) {
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
+ strerror (errno)));
+ return FALSE;
+ }
+
+ if (spec->width != 16)
+ return FALSE;
+
+ ports = ainfo.play.port;
+
+ AUDIO_INITINFO (&ainfo);
+
+ ainfo.play.sample_rate = spec->rate;
+ ainfo.play.channels = spec->channels;
+ ainfo.play.precision = spec->width;
+ ainfo.play.encoding = AUDIO_ENCODING_LINEAR;
+ ainfo.play.port = ports;
+
+ /* buffer_time for playback is not implemented in Solaris at the moment,
+ but at some point in the future, it might be */
+ ainfo.play.buffer_size =
+ gst_util_uint64_scale (spec->rate * spec->bytes_per_sample,
+ spec->buffer_time, GST_SECOND / GST_USECOND);
+
+ spec->silence_sample[0] = 0;
+ spec->silence_sample[1] = 0;
+ spec->silence_sample[2] = 0;
+ spec->silence_sample[3] = 0;
+
+ ret = ioctl (sunaudiosink->fd, AUDIO_SETINFO, &ainfo);
+ if (ret == -1) {
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
+ strerror (errno)));
+ return FALSE;
+ }
+
+ /* Now read back the info to find out the actual buffer size and set
+ segtotal */
+ AUDIO_INITINFO (&ainfo);
+
+ ret = ioctl (sunaudiosink->fd, AUDIO_GETINFO, &ainfo);
+ if (ret == -1) {
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
+ strerror (errno)));
+ return FALSE;
+ }
+#if 0
+ /* We don't actually use the buffer_size from the sound device, because
+ * it seems it's just bogus sometimes */
+ sunaudiosink->segtotal = spec->segtotal =
+ ainfo.play.buffer_size / spec->segsize;
+#else
+ sunaudiosink->segtotal = spec->segtotal;
+#endif
+ sunaudiosink->segtotal_samples =
+ spec->segtotal * spec->segsize / spec->bytes_per_sample;
+
+ sunaudiosink->segs_written = (gint) ainfo.play.eof;
+ sunaudiosink->samples_written = ainfo.play.samples;
+ sunaudiosink->bytes_per_sample = spec->bytes_per_sample;
+
+ GST_DEBUG_OBJECT (sunaudiosink, "Got device buffer_size of %u",
+ ainfo.play.buffer_size);
+
+ return TRUE;
+}
+
+static gboolean
+gst_sunaudiosink_unprepare (GstAudioSink * asink)
+{
+ return TRUE;
+}
+
+#define LOOP_WHILE_EINTR(v,func) do { (v) = (func); } \
+ while ((v) == -1 && errno == EINTR);
+
+/* Called with the write_mutex held */
+static void
+gst_sunaudio_sink_do_delay (GstSunAudioSink * sink)
+{
+ GstBaseAudioSink *ba_sink = GST_BASE_AUDIO_SINK (sink);
+ GstClockTime total_sleep;
+ GstClockTime max_sleep;
+ gint sleep_usecs;
+ GTimeVal sleep_end;
+ gint err;
+ audio_info_t ainfo;
+ guint diff;
+
+ /* This code below ensures that we don't race any further than buffer_time
+ * ahead of the audio output, by sleeping if the next write call would cause
+ * us to advance too far in the ring-buffer */
+ LOOP_WHILE_EINTR (err, ioctl (sink->fd, AUDIO_GETINFO, &ainfo));
+ if (err < 0)
+ goto write_error;
+
+ /* Compute our offset from the output (copes with overflow) */
+ diff = (guint) (sink->segs_written) - ainfo.play.eof;
+ if (diff > sink->segtotal) {
+ /* This implies that reset did a flush just as the sound device aquired
+ * some buffers internally, and it causes us to be out of sync with the
+ * eof measure. This corrects it */
+ sink->segs_written = ainfo.play.eof;
+ diff = 0;
+ }
+
+ if (diff + 1 < sink->segtotal)
+ return; /* no need to sleep at all */
+
+ /* Never sleep longer than the initial number of undrained segments in the
+ device plus one */
+ total_sleep = 0;
+ max_sleep = (diff + 1) * (ba_sink->latency_time * GST_USECOND);
+ /* sleep for a segment period between .eof polls */
+ sleep_usecs = ba_sink->latency_time;
+
+ /* Current time is our reference point */
+ g_get_current_time (&sleep_end);
+
+ /* If the next segment would take us too far along the ring buffer,
+ * sleep for a bit to free up a slot. If there were a way to find out
+ * when the eof field actually increments, we could use, but the only
+ * notification mechanism seems to be SIGPOLL, which we can't use from
+ * a support library */
+ while (diff + 1 >= sink->segtotal && total_sleep < max_sleep) {
+ GST_LOG_OBJECT (sink, "need to block to drain segment(s). "
+ "Sleeping for %d us", sleep_usecs);
+
+ g_time_val_add (&sleep_end, sleep_usecs);
+
+ if (g_cond_timed_wait (sink->sleep_cond, sink->write_mutex, &sleep_end)) {
+ GST_LOG_OBJECT (sink, "Waking up early due to reset");
+ return; /* Got told to wake up */
+ }
+ total_sleep += (sleep_usecs * GST_USECOND);
+
+ LOOP_WHILE_EINTR (err, ioctl (sink->fd, AUDIO_GETINFO, &ainfo));
+ if (err < 0)
+ goto write_error;
+
+ /* Compute our (new) offset from the output (copes with overflow) */
+ diff = (guint) g_atomic_int_get (&sink->segs_written) - ainfo.play.eof;
+ }
+
+ return;
+
+write_error:
+ GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
+ ("Playback error on device '%s': %s", sink->device, strerror (errno)));
+ return;
+}
+
+static guint
+gst_sunaudiosink_write (GstAudioSink * asink, gpointer data, guint length)
+{
+ GstSunAudioSink *sink = GST_SUNAUDIO_SINK (asink);
+
+ gint bytes_written, err;
+
+ g_mutex_lock (sink->write_mutex);
+ if (sink->flushing) {
+ /* Exit immediately if reset tells us to */
+ g_mutex_unlock (sink->write_mutex);
+ return length;
+ }
+
+ LOOP_WHILE_EINTR (bytes_written, write (sink->fd, data, length));
+ if (bytes_written < 0) {
+ err = bytes_written;
+ goto write_error;
+ }
+
+ /* Increment our sample counter, for delay calcs */
+ g_atomic_int_add (&sink->samples_written, length / sink->bytes_per_sample);
+
+ /* Don't consider the segment written if we didn't output the whole lot yet */
+ if (bytes_written < length) {
+ g_mutex_unlock (sink->write_mutex);
+ return (guint) bytes_written;
+ }
+
+ /* Write a zero length output to trigger increment of the eof field */
+ LOOP_WHILE_EINTR (err, write (sink->fd, NULL, 0));
+ if (err < 0)
+ goto write_error;
+
+ /* Count this extra segment we've written */
+ sink->segs_written += 1;
+
+ /* Now delay so we don't overrun the ring buffer */
+ gst_sunaudio_sink_do_delay (sink);
+
+ g_mutex_unlock (sink->write_mutex);
+ return length;
+
+write_error:
+ g_mutex_unlock (sink->write_mutex);
+
+ GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
+ ("Playback error on device '%s': %s", sink->device, strerror (errno)));
+ return length; /* Say we wrote the segment to let the ringbuffer exit */
+}
+
+/*
+ * Provide the current number of unplayed samples that have been written
+ * to the device */
+static guint
+gst_sunaudiosink_delay (GstAudioSink * asink)
+{
+ GstSunAudioSink *sink = GST_SUNAUDIO_SINK (asink);
+ audio_info_t ainfo;
+ gint ret;
+ guint offset;
+
+ ret = ioctl (sink->fd, AUDIO_GETINFO, &ainfo);
+ if (G_UNLIKELY (ret == -1))
+ return 0;
+
+ offset = (g_atomic_int_get (&sink->samples_written) - ainfo.play.samples);
+
+ /* If the offset is larger than the total ringbuffer size, then we asked
+ between the write call and when samples_written is updated */
+ if (G_UNLIKELY (offset > sink->segtotal_samples))
+ return 0;
+
+ return offset;
+}
+
+static void
+gst_sunaudiosink_reset (GstAudioSink * asink)
+{
+ /* Get current values */
+ GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
+ audio_info_t ainfo;
+ int ret;
+
+ ret = ioctl (sunaudiosink->fd, AUDIO_GETINFO, &ainfo);
+ if (ret == -1) {
+ /*
+ * Should never happen, but if we couldn't getinfo, then no point
+ * trying to setinfo
+ */
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
+ strerror (errno)));
+ return;
+ }
+
+ /*
+ * Pause the audio - so audio stops playing immediately rather than
+ * waiting for the ringbuffer to empty.
+ */
+ ainfo.play.pause = !NULL;
+ ret = ioctl (sunaudiosink->fd, AUDIO_SETINFO, &ainfo);
+ if (ret == -1) {
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
+ strerror (errno)));
+ }
+
+ /* Flush the audio */
+ ret = ioctl (sunaudiosink->fd, I_FLUSH, FLUSHW);
+ if (ret == -1) {
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
+ strerror (errno)));
+ }
+
+ /* Now, we take the write_mutex and signal to ensure the write thread
+ * is not busy, and we signal the condition to wake up any sleeper,
+ * then we flush again in case the write wrote something after we flushed,
+ * and finally release the lock and unpause */
+ g_mutex_lock (sunaudiosink->write_mutex);
+ sunaudiosink->flushing = TRUE;
+
+ g_cond_signal (sunaudiosink->sleep_cond);
+
+ ret = ioctl (sunaudiosink->fd, I_FLUSH, FLUSHW);
+ if (ret == -1) {
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
+ strerror (errno)));
+ }
+
+ /* unpause the audio */
+ ainfo.play.pause = NULL;
+ ret = ioctl (sunaudiosink->fd, AUDIO_SETINFO, &ainfo);
+ if (ret == -1) {
+ GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
+ strerror (errno)));
+ }
+
+ /* After flushing the audio device, we need to remeasure the sample count
+ * and segments written count so we're in sync with the device */
+
+ sunaudiosink->segs_written = ainfo.play.eof;
+ g_atomic_int_set (&sunaudiosink->samples_written, ainfo.play.samples);
+
+ sunaudiosink->flushing = FALSE;
+ g_mutex_unlock (sunaudiosink->write_mutex);
+}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-21 21:01:50 +0000