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Diffstat (limited to 'sys/oss/gstosssrc.c')
-rw-r--r--sys/oss/gstosssrc.c537
1 files changed, 537 insertions, 0 deletions
diff --git a/sys/oss/gstosssrc.c b/sys/oss/gstosssrc.c
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+++ b/sys/oss/gstosssrc.c
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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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-osssrc
+ *
+ * This element lets you record sound using the Open Sound System (OSS).
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -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 "gstosssrc.h"
+#include "common.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,
+};
+
+GST_BOILERPLATE_WITH_INTERFACE (GstOssSrc, gst_oss_src, GstAudioSrc,
+ GST_TYPE_AUDIO_SRC, GstMixer, GST_TYPE_MIXER, gst_oss_src_mixer);
+
+GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssSrc, gst_oss_src_mixer);
+
+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);
+
+static gboolean gst_oss_src_open (GstAudioSrc * asrc);
+static gboolean gst_oss_src_close (GstAudioSrc * asrc);
+static gboolean gst_oss_src_prepare (GstAudioSrc * asrc,
+ GstRingBufferSpec * spec);
+static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc);
+static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length);
+static guint gst_oss_src_delay (GstAudioSrc * asrc);
+static void gst_oss_src_reset (GstAudioSrc * asrc);
+
+
+
+static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
+ "signed = (boolean) { TRUE, FALSE }, "
+ "width = (int) 16, "
+ "depth = (int) 16, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
+ "audio/x-raw-int, "
+ "signed = (boolean) { TRUE, FALSE }, "
+ "width = (int) 8, "
+ "depth = (int) 8, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
+ );
+
+
+static void
+gst_oss_src_dispose (GObject * object)
+{
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_oss_src_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+ gst_element_class_set_details_simple (element_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_static_pad_template (element_class,
+ &osssrc_src_factory);
+}
+
+static void
+gst_oss_src_class_init (GstOssSrcClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstBaseSrcClass *gstbasesrc_class;
+ GstAudioSrcClass *gstaudiosrc_class;
+
+ gobject_class = (GObjectClass *) 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));
+}
+
+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:
+ if (src->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, GstOssSrcClass * g_class)
+{
+ 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)
+{
+ 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);
+
+ 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 (GstBufferFormat fmt)
+{
+ gint result;
+
+ switch (fmt) {
+ case GST_MU_LAW:
+ result = AFMT_MU_LAW;
+ break;
+ case GST_A_LAW:
+ result = AFMT_A_LAW;
+ break;
+ case GST_IMA_ADPCM:
+ result = AFMT_IMA_ADPCM;
+ break;
+ case GST_U8:
+ result = AFMT_U8;
+ break;
+ case GST_S16_LE:
+ result = AFMT_S16_LE;
+ break;
+ case GST_S16_BE:
+ result = AFMT_S16_BE;
+ break;
+ case GST_S8:
+ result = AFMT_S8;
+ break;
+ case GST_U16_LE:
+ result = AFMT_U16_LE;
+ break;
+ case GST_U16_BE:
+ result = AFMT_U16_BE;
+ break;
+ case GST_MPEG:
+ result = AFMT_MPEG;
+ 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;
+ }
+ }
+
+ if (!oss->mixer) {
+ oss->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_CAPTURE);
+
+ if (oss->mixer) {
+ g_free (oss->device_name);
+ oss->device_name = g_strdup (oss->mixer->cardname);
+ }
+ }
+ 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);
+
+ if (oss->mixer) {
+ gst_ossmixer_free (oss->mixer);
+ oss->mixer = NULL;
+ }
+
+ gst_caps_replace (&oss->probed_caps, NULL);
+
+ return TRUE;
+}
+
+static gboolean
+gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
+{
+ GstOssSrc *oss;
+ struct audio_buf_info info;
+ int mode;
+ int fmt, tmp;
+
+ 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->format);
+ if (fmt == 0)
+ goto wrong_format;
+
+ 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 (spec->channels == 2)
+ SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
+ SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS");
+ SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED");
+
+ GET_PARAM (oss, SNDCTL_DSP_GETISPACE, &info, "GETISPACE");
+
+ spec->segsize = info.fragsize;
+ spec->segtotal = info.fragstotal;
+
+ if (spec->width != 16 && spec->width != 8)
+ goto dodgy_width;
+
+ spec->bytes_per_sample = (spec->width / 8) * spec->channels;
+ oss->bytes_per_sample = (spec->width / 8) * spec->channels;
+ memset (spec->silence_sample, 0, spec->bytes_per_sample);
+
+ 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", spec->format), (NULL));
+ return FALSE;
+ }
+dodgy_width:
+ {
+ GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
+ ("Unexpected width %d", spec->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)
+{
+ 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 */
+}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-21 16:42:12 +0000