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gstreamer/sys/oss/gstosssrc.c
Reynaldo H. Verdejo Pinochet 492e917a6d osssrc: remove unnecessary NULL check before g_free()
2015-11-15 01:43:08 -08:00

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
* gstosssrc.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-osssrc
*
* This element lets you record sound using the Open Sound System (OSS).
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 -v osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
* ]| will record sound from your sound card using OSS and encode it to an
* Ogg/Vorbis file (this will only work if your mixer settings are right
* and the right inputs enabled etc.)
* </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 "gstosssrc.h"
#include <gst/gst-i18n-plugin.h>
GST_DEBUG_CATEGORY_EXTERN (oss_debug);
#define GST_CAT_DEFAULT oss_debug
#define DEFAULT_DEVICE "/dev/dsp"
#define DEFAULT_DEVICE_NAME ""
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME,
};
#define gst_oss_src_parent_class parent_class
G_DEFINE_TYPE (GstOssSrc, gst_oss_src, GST_TYPE_AUDIO_SRC);
static void gst_oss_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_oss_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_oss_src_dispose (GObject * object);
static void gst_oss_src_finalize (GstOssSrc * osssrc);
static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter);
static gboolean gst_oss_src_open (GstAudioSrc * asrc);
static gboolean gst_oss_src_close (GstAudioSrc * asrc);
static gboolean gst_oss_src_prepare (GstAudioSrc * asrc,
GstAudioRingBufferSpec * spec);
static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc);
static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp);
static guint gst_oss_src_delay (GstAudioSrc * asrc);
static void gst_oss_src_reset (GstAudioSrc * asrc);
#define FORMATS "{" GST_AUDIO_NE(S16)","GST_AUDIO_NE(U16)", S8, U8 }"
static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
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 void
gst_oss_src_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_oss_src_class_init (GstOssSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstaudiosrc_class = (GstAudioSrcClass *) klass;
gobject_class->dispose = gst_oss_src_dispose;
gobject_class->finalize = (GObjectFinalizeFunc) gst_oss_src_finalize;
gobject_class->get_property = gst_oss_src_get_property;
gobject_class->set_property = gst_oss_src_set_property;
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_src_getcaps);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss_src_open);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_src_prepare);
gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_src_unprepare);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss_src_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss_src_read);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss_src_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss_src_reset);
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));
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));
gst_element_class_set_static_metadata (gstelement_class, "Audio Source (OSS)",
"Source/Audio",
"Capture from a sound card via OSS",
"Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&osssrc_src_factory));
}
static void
gst_oss_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstOssSrc *src;
src = GST_OSS_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_free (src->device);
src->device = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_oss_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstOssSrc *src;
src = GST_OSS_SRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, src->device);
break;
case PROP_DEVICE_NAME:
g_value_set_string (value, src->device_name);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_oss_src_init (GstOssSrc * osssrc)
{
const gchar *device;
GST_DEBUG ("initializing osssrc");
device = g_getenv ("AUDIODEV");
if (device == NULL)
device = DEFAULT_DEVICE;
osssrc->fd = -1;
osssrc->device = g_strdup (device);
osssrc->device_name = g_strdup (DEFAULT_DEVICE_NAME);
osssrc->probed_caps = NULL;
}
static void
gst_oss_src_finalize (GstOssSrc * osssrc)
{
g_free (osssrc->device);
g_free (osssrc->device_name);
G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (osssrc));
}
static GstCaps *
gst_oss_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstOssSrc *osssrc;
GstCaps *caps;
osssrc = GST_OSS_SRC (bsrc);
if (osssrc->fd == -1) {
GST_DEBUG_OBJECT (osssrc, "device not open, using template caps");
return NULL; /* base class will get template caps for us */
}
if (osssrc->probed_caps) {
GST_LOG_OBJECT (osssrc, "Returning cached caps");
return gst_caps_ref (osssrc->probed_caps);
}
caps = gst_oss_helper_probe_caps (osssrc->fd);
if (caps) {
osssrc->probed_caps = gst_caps_ref (caps);
}
GST_INFO_OBJECT (osssrc, "returning caps %" GST_PTR_FORMAT, 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_src_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_src_open (GstAudioSrc * asrc)
{
GstOssSrc *oss;
int mode;
oss = GST_OSS_SRC (asrc);
mode = O_RDONLY;
mode |= O_NONBLOCK;
oss->fd = open (oss->device, mode, 0);
if (oss->fd == -1) {
switch (errno) {
case EACCES:
goto no_permission;
default:
goto open_failed;
}
}
g_free (oss->device_name);
oss->device_name = gst_oss_helper_get_card_name ("/dev/mixer");
return TRUE;
no_permission:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
(_("Could not open audio device for recording. "
"You don't have permission to open the device.")),
GST_ERROR_SYSTEM);
return FALSE;
}
open_failed:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
(_("Could not open audio device for recording.")),
("Unable to open device %s for recording: %s",
oss->device, g_strerror (errno)));
return FALSE;
}
}
static gboolean
gst_oss_src_close (GstAudioSrc * asrc)
{
GstOssSrc *oss;
oss = GST_OSS_SRC (asrc);
close (oss->fd);
gst_caps_replace (&oss->probed_caps, NULL);
return TRUE;
}
static gboolean
gst_oss_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstOssSrc *oss;
struct audio_buf_info info;
int mode;
int fmt, tmp;
guint width, rate, channels;
oss = GST_OSS_SRC (asrc);
mode = fcntl (oss->fd, F_GETFL);
mode &= ~O_NONBLOCK;
if (fcntl (oss->fd, F_SETFL, mode) == -1)
goto non_block;
fmt = gst_oss_src_get_format (spec->type,
GST_AUDIO_INFO_FORMAT (&spec->info));
if (fmt == 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;
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");
SET_PARAM (oss, SNDCTL_DSP_RESET, 0, "RESET");
SET_PARAM (oss, SNDCTL_DSP_SETFMT, fmt, "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");
GET_PARAM (oss, SNDCTL_DSP_GETISPACE, &info, "GETISPACE");
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;
non_block:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
("Unable to set device %s in non blocking mode: %s",
oss->device, g_strerror (errno)), (NULL));
return FALSE;
}
wrong_format:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
("Unable to get format (%d, %d)", spec->type,
GST_AUDIO_INFO_FORMAT (&spec->info)), (NULL));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
("Unexpected width %d", width), (NULL));
return FALSE;
}
}
static gboolean
gst_oss_src_unprepare (GstAudioSrc * asrc)
{
/* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
if (!gst_oss_src_close (asrc))
goto couldnt_close;
if (!gst_oss_src_open (asrc))
goto couldnt_reopen;
return TRUE;
couldnt_close:
{
GST_DEBUG_OBJECT (asrc, "Could not close the audio device");
return FALSE;
}
couldnt_reopen:
{
GST_DEBUG_OBJECT (asrc, "Could not reopen the audio device");
return FALSE;
}
}
static guint
gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{
return read (GST_OSS_SRC (asrc)->fd, data, length);
}
static guint
gst_oss_src_delay (GstAudioSrc * asrc)
{
GstOssSrc *oss;
gint delay = 0;
gint ret;
oss = GST_OSS_SRC (asrc);
#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_src_reset (GstAudioSrc * asrc)
{
/* 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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