gstreamer/sys/sunaudio/gstsunaudiosink.c
Brian Cameron 23b9485530 sys/sunaudio/: Fix up copyrights (#525860).
Original commit message from CVS:
Patch by: Brian Cameron <brian.cameron at sun dot com>
* sys/sunaudio/gstsunaudio.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiomixer.h:
* sys/sunaudio/gstsunaudiomixerctrl.c:
* sys/sunaudio/gstsunaudiomixerctrl.h:
* sys/sunaudio/gstsunaudiomixertrack.c:
* sys/sunaudio/gstsunaudiomixertrack.h:
* sys/sunaudio/gstsunaudiosink.c:
* sys/sunaudio/gstsunaudiosink.h:
* sys/sunaudio/gstsunaudiosrc.c:
* sys/sunaudio/gstsunaudiosrc.h:
Fix up copyrights (#525860).
2008-04-02 22:37:29 +00:00

665 lines
20 KiB
C

/*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-sunaudiosink
*
* <refsect2>
* <para>
* sunaudiosink is an audio sink designed to work with the Sun Audio
* interface available in Solaris.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch -v sinesrc ! sunaudiosink
* </programlisting>
* </para>
* </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
/* elementfactory information */
static const GstElementDetails plugin_details =
GST_ELEMENT_DETAILS ("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_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_details (element_class, &plugin_details);
}
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_DEBUG_FUNCPTR (gst_sunaudiosink_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (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));
}
static void
gst_sunaudiosink_init (GstSunAudioSink * sunaudiosink)
{
GstBaseAudioSink *ba_sink = GST_BASE_AUDIO_SINK (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 gconf 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;
poll_failed:
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);
}