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gstreamer/sys/oss/gstosssink.c

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
* gstosssink.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-osssink
*
* This element lets you output sound using the Open Sound System (OSS).
*
* Note that you should almost always use generic audio conversion elements
* like audioconvert and audioresample in front of an audiosink to make sure
* your pipeline works under all circumstances (those conversion elements will
* act in passthrough-mode if no conversion is necessary).
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.1 ! osssink
* ]| will output a sine wave (continuous beep sound) to your sound card (with
* a very low volume as precaution).
* |[
* gst-launch-1.0 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! osssink
* ]| will play an Ogg/Vorbis audio file and output it using the Open Sound System.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <sys/ioctl.h>
#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
#ifdef HAVE_OSS_INCLUDE_IN_SYS
# include <sys/soundcard.h>
#else
# ifdef HAVE_OSS_INCLUDE_IN_ROOT
# include <soundcard.h>
# else
# ifdef HAVE_OSS_INCLUDE_IN_MACHINE
# include <machine/soundcard.h>
# else
# error "What to include?"
# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
#endif /* HAVE_OSS_INCLUDE_IN_SYS */
#include "common.h"
#include "gstosssink.h"
#include <gst/gst-i18n-plugin.h>
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
static void gst_oss_sink_dispose (GObject * object);
static void gst_oss_sink_finalise (GObject * object);
static void gst_oss_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_oss_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink, GstCaps * filter);
static gboolean gst_oss_sink_open (GstAudioSink * asink);
static gboolean gst_oss_sink_close (GstAudioSink * asink);
static gboolean gst_oss_sink_prepare (GstAudioSink * asink,
GstAudioRingBufferSpec * spec);
static gboolean gst_oss_sink_unprepare (GstAudioSink * asink);
static gint gst_oss_sink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_oss_sink_delay (GstAudioSink * asink);
static void gst_oss_sink_reset (GstAudioSink * asink);
/* OssSink signals and args */
enum
{
LAST_SIGNAL
};
#define DEFAULT_DEVICE "/dev/dsp"
enum
{
PROP_0,
PROP_DEVICE,
};
#define FORMATS "{" GST_AUDIO_NE(S16)","GST_AUDIO_NE(U16)", S8, U8 }"
static GstStaticPadTemplate osssink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " FORMATS ", "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], "
"channels = (int) 1; "
"audio/x-raw, "
"format = (string) " FORMATS ", "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], "
"channels = (int) 2, " "channel-mask = (bitmask) 0x3")
);
/* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */
#define gst_oss_sink_parent_class parent_class
G_DEFINE_TYPE (GstOssSink, gst_oss_sink, GST_TYPE_AUDIO_SINK);
static void
gst_oss_sink_dispose (GObject * object)
{
GstOssSink *osssink = GST_OSSSINK (object);
if (osssink->probed_caps) {
gst_caps_unref (osssink->probed_caps);
osssink->probed_caps = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_oss_sink_class_init (GstOssSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstAudioSinkClass *gstaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstaudiosink_class = (GstAudioSinkClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = gst_oss_sink_dispose;
gobject_class->finalize = gst_oss_sink_finalise;
gobject_class->get_property = gst_oss_sink_get_property;
gobject_class->set_property = gst_oss_sink_set_property;
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"OSS device (usually /dev/dspN)", DEFAULT_DEVICE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare);
gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset);
gst_element_class_set_static_metadata (gstelement_class, "Audio Sink (OSS)",
"Sink/Audio",
"Output to a sound card via OSS",
"Erik Walthinsen <omega@cse.ogi.edu>, "
"Wim Taymans <wim.taymans@chello.be>");
gst_element_class_add_static_pad_template (gstelement_class,
&osssink_sink_factory);
}
static void
gst_oss_sink_init (GstOssSink * osssink)
{
const gchar *device;
GST_DEBUG_OBJECT (osssink, "initializing osssink");
device = g_getenv ("AUDIODEV");
if (device == NULL)
device = DEFAULT_DEVICE;
osssink->device = g_strdup (device);
osssink->fd = -1;
}
static void
gst_oss_sink_finalise (GObject * object)
{
GstOssSink *osssink = GST_OSSSINK (object);
g_free (osssink->device);
G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (object));
}
static void
gst_oss_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOssSink *sink;
sink = GST_OSSSINK (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (sink->device);
sink->device = g_value_dup_string (value);
if (sink->probed_caps) {
gst_caps_unref (sink->probed_caps);
sink->probed_caps = NULL;
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_oss_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOssSink *sink;
sink = GST_OSSSINK (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, sink->device);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_oss_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
{
GstOssSink *osssink;
GstCaps *caps;
osssink = GST_OSSSINK (bsink);
if (osssink->fd == -1) {
caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink));
} else if (osssink->probed_caps) {
caps = gst_caps_ref (osssink->probed_caps);
} else {
caps = gst_oss_helper_probe_caps (osssink->fd);
if (caps && !gst_caps_is_empty (caps)) {
osssink->probed_caps = gst_caps_ref (caps);
}
}
if (filter && caps) {
GstCaps *intersection;
intersection =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
return intersection;
} else {
return caps;
}
}
static gint
ilog2 (gint x)
{
/* well... hacker's delight explains... */
x = x | (x >> 1);
x = x | (x >> 2);
x = x | (x >> 4);
x = x | (x >> 8);
x = x | (x >> 16);
x = x - ((x >> 1) & 0x55555555);
x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
x = (x + (x >> 4)) & 0x0f0f0f0f;
x = x + (x >> 8);
x = x + (x >> 16);
return (x & 0x0000003f) - 1;
}
static gint
gst_oss_sink_get_format (GstAudioRingBufferFormatType fmt, GstAudioFormat rfmt)
{
gint result;
switch (fmt) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
result = AFMT_MU_LAW;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
result = AFMT_A_LAW;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM:
result = AFMT_IMA_ADPCM;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
result = AFMT_MPEG;
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
{
switch (rfmt) {
case GST_AUDIO_FORMAT_U8:
result = AFMT_U8;
break;
case GST_AUDIO_FORMAT_S16LE:
result = AFMT_S16_LE;
break;
case GST_AUDIO_FORMAT_S16BE:
result = AFMT_S16_BE;
break;
case GST_AUDIO_FORMAT_S8:
result = AFMT_S8;
break;
case GST_AUDIO_FORMAT_U16LE:
result = AFMT_U16_LE;
break;
case GST_AUDIO_FORMAT_U16BE:
result = AFMT_U16_BE;
break;
default:
result = 0;
break;
}
break;
}
default:
result = 0;
break;
}
return result;
}
static gboolean
gst_oss_sink_open (GstAudioSink * asink)
{
GstOssSink *oss;
int mode;
oss = GST_OSSSINK (asink);
mode = O_WRONLY;
mode |= O_NONBLOCK;
oss->fd = open (oss->device, mode, 0);
if (oss->fd == -1) {
switch (errno) {
case EBUSY:
goto busy;
case EACCES:
goto no_permission;
default:
goto open_failed;
}
}
return TRUE;
/* ERRORS */
busy:
{
GST_ELEMENT_ERROR (oss, RESOURCE, BUSY,
(_("Could not open audio device for playback. "
"Device is being used by another application.")), (NULL));
return FALSE;
}
no_permission:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
(_("Could not open audio device for playback. "
"You don't have permission to open the device.")),
GST_ERROR_SYSTEM);
return FALSE;
}
open_failed:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,
(_("Could not open audio device for playback.")), GST_ERROR_SYSTEM);
return FALSE;
}
}
static gboolean
gst_oss_sink_close (GstAudioSink * asink)
{
close (GST_OSSSINK (asink)->fd);
GST_OSSSINK (asink)->fd = -1;
return TRUE;
}
static gboolean
gst_oss_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
{
GstOssSink *oss;
struct audio_buf_info info;
int mode;
int tmp;
guint width, rate, channels;
oss = GST_OSSSINK (asink);
/* we opened non-blocking so that we can detect if the device is available
* without hanging forever. We now want to remove the non-blocking flag. */
mode = fcntl (oss->fd, F_GETFL);
mode &= ~O_NONBLOCK;
if (fcntl (oss->fd, F_SETFL, mode) == -1) {
/* some drivers do no support unsetting the non-blocking flag, try to
* close/open the device then. This is racy but we error out properly. */
gst_oss_sink_close (asink);
if ((oss->fd = open (oss->device, O_WRONLY, 0)) == -1)
goto non_block;
}
tmp = gst_oss_sink_get_format (spec->type,
GST_AUDIO_INFO_FORMAT (&spec->info));
if (tmp == 0)
goto wrong_format;
width = GST_AUDIO_INFO_WIDTH (&spec->info);
rate = GST_AUDIO_INFO_RATE (&spec->info);
channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
if (width != 16 && width != 8)
goto dodgy_width;
SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp, "SETFMT");
if (channels == 2)
SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
SET_PARAM (oss, SNDCTL_DSP_CHANNELS, channels, "CHANNELS");
SET_PARAM (oss, SNDCTL_DSP_SPEED, rate, "SPEED");
tmp = ilog2 (spec->segsize);
tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",
spec->segsize, spec->segtotal, tmp);
SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT");
GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info, "GETOSPACE");
spec->segsize = info.fragsize;
spec->segtotal = info.fragstotal;
oss->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
spec->segsize, spec->segtotal, tmp);
return TRUE;
/* ERRORS */
non_block:
{
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
("Unable to set device %s in non blocking mode: %s",
oss->device, g_strerror (errno)));
return FALSE;
}
wrong_format:
{
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
("Unable to get format (%d, %d)", spec->type,
GST_AUDIO_INFO_FORMAT (&spec->info)));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
("unexpected width %d", width));
return FALSE;
}
}
static gboolean
gst_oss_sink_unprepare (GstAudioSink * asink)
{
/* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
if (!gst_oss_sink_close (asink))
goto couldnt_close;
if (!gst_oss_sink_open (asink))
goto couldnt_reopen;
return TRUE;
/* ERRORS */
couldnt_close:
{
GST_DEBUG_OBJECT (asink, "Could not close the audio device");
return FALSE;
}
couldnt_reopen:
{
GST_DEBUG_OBJECT (asink, "Could not reopen the audio device");
return FALSE;
}
}
static gint
gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length)
{
return write (GST_OSSSINK (asink)->fd, data, length);
}
static guint
gst_oss_sink_delay (GstAudioSink * asink)
{
GstOssSink *oss;
gint delay = 0;
gint ret;
oss = GST_OSSSINK (asink);
#ifdef SNDCTL_DSP_GETODELAY
ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
#else
ret = -1;
#endif
if (ret < 0) {
audio_buf_info info;
ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
}
return delay / oss->bytes_per_sample;
}
static void
gst_oss_sink_reset (GstAudioSink * asink)
{
/* There's nothing we can do here really: OSS can't handle access to the
* same device/fd from multiple threads and might deadlock or blow up in
* other ways if we try an ioctl SNDCTL_DSP_RESET or similar */
}
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