Commit 6e9ee0d1 authored by Jan Schmidt's avatar Jan Schmidt
Browse files

sys/sunaudio/: Use the sunaudio debug category.

Original commit message from CVS:
* sys/sunaudio/gstsunaudiomixerctrl.c:
* sys/sunaudio/gstsunaudiosrc.c:
Use the sunaudio debug category.
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_finalize),
(gst_sunaudiosink_class_init), (gst_sunaudiosink_init),
(gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property),
(gst_sunaudiosink_open), (gst_sunaudiosink_close),
(gst_sunaudiosink_prepare), (gst_sunaudio_sink_do_delay),
(gst_sunaudiosink_write), (gst_sunaudiosink_delay),
(gst_sunaudiosink_reset):
* sys/sunaudio/gstsunaudiosink.h:
Uses the sunaudio debug category for all debug output
Implements the _delay() callback to synchronise video playback better
Change the segtotal and segsize values back to the parent class
defaults (taken from buffer_time and latency_times of 200ms and 10ms
respectively)
Measure the samples written to the device vs. played.
Keep track of segments in the device by writing empty eof frames, and
sleep using a GCond when we get too far ahead and risk overrunning the
sink's ringbuffer.
Fixes: #360673
parent f3bdb649
2006-12-08 Jan Schmidt <thaytan@mad.scientist.com>
* sys/sunaudio/gstsunaudiomixerctrl.c:
* sys/sunaudio/gstsunaudiosrc.c:
Use the sunaudio debug category.
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_finalize),
(gst_sunaudiosink_class_init), (gst_sunaudiosink_init),
(gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property),
(gst_sunaudiosink_open), (gst_sunaudiosink_close),
(gst_sunaudiosink_prepare), (gst_sunaudio_sink_do_delay),
(gst_sunaudiosink_write), (gst_sunaudiosink_delay),
(gst_sunaudiosink_reset):
* sys/sunaudio/gstsunaudiosink.h:
Uses the sunaudio debug category for all debug output
Implements the _delay() callback to synchronise video playback better
Change the segtotal and segsize values back to the parent class
defaults (taken from buffer_time and latency_times of 200ms and 10ms
respectively)
Measure the samples written to the device vs. played.
Keep track of segments in the device by writing empty eof frames, and
sleep using a GCond when we get too far ahead and risk overrunning the
sink's ringbuffer.
Fixes: #360673
2006-12-08 Wim Taymans <wim@fluendo.com>
Patch by: Sebastian Dröge <mail at slomosnail de >
......
......@@ -35,6 +35,9 @@
#include "gstsunaudiomixerctrl.h"
#include "gstsunaudiomixertrack.h"
GST_DEBUG_CATEGORY_EXTERN (sunaudio_debug);
#define GST_CAT_DEFAULT sunaudio_debug
#define SCALE_FACTOR 2.55 /* 255/100 */
static gboolean
......
......@@ -48,6 +48,9 @@
#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",
......@@ -60,6 +63,7 @@ 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);
......@@ -128,6 +132,24 @@ 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)
{
......@@ -156,6 +178,8 @@ gst_sunaudiosink_class_init (GstSunAudioSinkClass * 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 =
......@@ -175,12 +199,12 @@ gst_sunaudiosink_class_init (GstSunAudioSinkClass * klass)
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");
......@@ -191,6 +215,10 @@ gst_sunaudiosink_init (GstSunAudioSink * sunaudiosink)
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
......@@ -203,8 +231,10 @@ gst_sunaudiosink_set_property (GObject * object, guint prop_id,
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);
......@@ -222,7 +252,9 @@ gst_sunaudiosink_get_property (GObject * object, guint prop_id,
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);
......@@ -254,6 +286,7 @@ gst_sunaudiosink_open (GstAudioSink * 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) {
......@@ -262,31 +295,24 @@ gst_sunaudiosink_open (GstAudioSink * asink)
}
if (fd == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, OPEN_WRITE, (NULL),
("can't open connection to Sun Audio device %s", sunaudiosink->device));
return FALSE;
GST_OBJECT_UNLOCK (sunaudiosink);
goto open_failed;
}
sunaudiosink->fd = fd;
GST_OBJECT_UNLOCK (sunaudiosink);
ret = ioctl (fd, AUDIO_GETDEV, &sunaudiosink->dev);
if (ret == -1) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
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) {
GST_ELEMENT_ERROR (sunaudiosink, RESOURCE, SETTINGS, (NULL), ("%s",
strerror (errno)));
return FALSE;
}
if (ret == -1)
goto ioctl_error;
GST_DEBUG_OBJECT (sunaudiosink, "monitor_gain %d",
sunaudiosink->info.monitor_gain);
......@@ -300,6 +326,16 @@ gst_sunaudiosink_open (GstAudioSink * asink)
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
......@@ -307,8 +343,10 @@ gst_sunaudiosink_close (GstAudioSink * asink)
{
GstSunAudioSink *sunaudiosink = GST_SUNAUDIO_SINK (asink);
close (sunaudiosink->fd);
sunaudiosink->fd = -1;
if (sunaudiosink->fd != -1) {
close (sunaudiosink->fd);
sunaudiosink->fd = -1;
}
return TRUE;
}
......@@ -340,16 +378,12 @@ gst_sunaudiosink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
ainfo.play.encoding = AUDIO_ENCODING_LINEAR;
ainfo.play.port = ports;
/*
* SunAudio doesn't really give access to buffer size, these values work. Setting
* the buffer so large (512K) is a bit annoying because this causes the volume
* control in audio players to be slow in responding since the audio volume won't
* change until the buffer empties. SunAudio doesn't seem to allow changing the
* audio output buffer size to anything smaller, though. I notice setting the
* values smaller causes the audio to stutter, which is worse.
*/
spec->segsize = 4096;
spec->segtotal = 128;
/* 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;
......@@ -362,6 +396,28 @@ gst_sunaudiosink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
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;
}
sunaudiosink->segtotal = spec->segtotal =
ainfo.play.buffer_size / spec->segsize;
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;
}
......@@ -371,21 +427,163 @@ 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)
{
return write (GST_SUNAUDIO_SINK (asink)->fd, data, 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 */
}
/*
* Should provide the current delay between writing a sample to the
* audio device and that sample being actually played. Returning 0 for
* now, but this isn't good for synchronization
*/
* Provide the current number of unplayed samples that have been written
* to the device */
static guint
gst_sunaudiosink_delay (GstAudioSink * asink)
{
return 0;
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
......@@ -425,6 +623,21 @@ gst_sunaudiosink_reset (GstAudioSink * asink)
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);
......@@ -432,4 +645,13 @@ gst_sunaudiosink_reset (GstAudioSink * asink)
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);
}
......@@ -47,7 +47,20 @@ struct _GstSunAudioSink {
audio_device_t dev;
audio_info_t info;
gint bytes_per_sample;
/* Number of segments the ringbuffer is configured for */
guint segtotal;
guint segtotal_samples;
/* Number of segments written to the device */
gint segs_written;
/* Number of samples written to the device */
gint samples_written;
guint bytes_per_sample;
/* mutex and gconf used to control the write method */
GMutex *write_mutex;
GCond *sleep_cond;
gboolean flushing;
};
struct _GstSunAudioSinkClass {
......
......@@ -50,6 +50,9 @@
#include "gstsunaudiosrc.h"
GST_DEBUG_CATEGORY_EXTERN (sunaudio_debug);
#define GST_CAT_DEFAULT sunaudio_debug
static GstElementDetails plugin_details =
GST_ELEMENT_DETAILS ("Sun Audio Source",
"Source/Audio",
......
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