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b603c27ea0
Original commit message from CVS: * sys/oss4/oss4-mixer.c: * sys/oss4/oss4-sink.c: * sys/oss4/oss4-source.c: Add some spaces in translateable strings. Fixes: #555969 #555968 #555965
1006 lines
28 KiB
C
1006 lines
28 KiB
C
/* GStreamer OSS4 audio source
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* Copyright (C) 2007-2008 Tim-Philipp Müller <tim centricular net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-oss4src
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*
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* This element lets you record sound using the Open Sound System (OSS)
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* version 4.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch -v oss4src ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
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* ]| will record sound from your sound card using OSS4 and encode it to an
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* Ogg/Vorbis file (this will only work if your mixer settings are right
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* and the right inputs areenabled etc.)
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* </refsect2>
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*
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* Since: 0.10.7
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*/
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/* FIXME: make sure we're not doing ioctls from the app thread (e.g. via the
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* mixer interface) while recording */
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <sys/types.h>
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#include <sys/stat.h>
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#include <sys/ioctl.h>
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#include <fcntl.h>
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#include <errno.h>
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#include <unistd.h>
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#include <string.h>
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#include <gst/interfaces/mixer.h>
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#include <gst/gst-i18n-plugin.h>
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#define NO_LEGACY_MIXER
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#include "oss4-audio.h"
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#include "oss4-source.h"
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#include "oss4-property-probe.h"
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#include "oss4-soundcard.h"
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#define GST_OSS4_SOURCE_IS_OPEN(src) (GST_OSS4_SOURCE(src)->fd != -1)
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GST_DEBUG_CATEGORY_EXTERN (oss4src_debug);
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#define GST_CAT_DEFAULT oss4src_debug
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#define DEFAULT_DEVICE NULL
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#define DEFAULT_DEVICE_NAME NULL
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enum
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{
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PROP_0,
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PROP_DEVICE,
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PROP_DEVICE_NAME
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};
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static void gst_oss4_source_init_interfaces (GType type);
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GST_BOILERPLATE_FULL (GstOss4Source, gst_oss4_source, GstAudioSrc,
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GST_TYPE_AUDIO_SRC, gst_oss4_source_init_interfaces);
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static void gst_oss4_source_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_oss4_source_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_oss4_source_dispose (GObject * object);
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static void gst_oss4_source_finalize (GstOss4Source * osssrc);
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static GstCaps *gst_oss4_source_getcaps (GstBaseSrc * bsrc);
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static gboolean gst_oss4_source_open (GstAudioSrc * asrc,
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gboolean silent_errors);
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static gboolean gst_oss4_source_open_func (GstAudioSrc * asrc);
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static gboolean gst_oss4_source_close (GstAudioSrc * asrc);
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static gboolean gst_oss4_source_prepare (GstAudioSrc * asrc,
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GstRingBufferSpec * spec);
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static gboolean gst_oss4_source_unprepare (GstAudioSrc * asrc);
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static guint gst_oss4_source_read (GstAudioSrc * asrc, gpointer data,
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guint length);
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static guint gst_oss4_source_delay (GstAudioSrc * asrc);
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static void gst_oss4_source_reset (GstAudioSrc * asrc);
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static void
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gst_oss4_source_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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GstPadTemplate *templ;
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gst_element_class_set_details_simple (element_class,
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"OSS v4 Audio Source", "Source/Audio",
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"Capture from a sound card via OSS version 4",
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"Tim-Philipp Müller <tim centricular net>");
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templ = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
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gst_oss4_audio_get_template_caps ());
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gst_element_class_add_pad_template (element_class, templ);
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}
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static void
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gst_oss4_source_class_init (GstOss4SourceClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSrcClass *gstbasesrc_class;
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GstBaseAudioSrcClass *gstbaseaudiosrc_class;
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GstAudioSrcClass *gstaudiosrc_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesrc_class = (GstBaseSrcClass *) klass;
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gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
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gstaudiosrc_class = (GstAudioSrcClass *) klass;
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss4_source_dispose);
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gobject_class->finalize =
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(GObjectFinalizeFunc) GST_DEBUG_FUNCPTR (gst_oss4_source_finalize);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_oss4_source_get_property);
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_oss4_source_set_property);
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss4_source_getcaps);
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gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss4_source_open_func);
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gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss4_source_prepare);
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gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss4_source_unprepare);
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gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss4_source_close);
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gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss4_source_read);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss4_source_delay);
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gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss4_source_reset);
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g_object_class_install_property (gobject_class, PROP_DEVICE,
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g_param_spec_string ("device", "Device",
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"OSS4 device (e.g. /dev/oss/hdaudio0/pcm0 or /dev/dspN) "
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"(NULL = use first available device)",
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DEFAULT_DEVICE, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
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g_param_spec_string ("device-name", "Device name",
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"Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
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G_PARAM_READABLE));
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}
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static void
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gst_oss4_source_init (GstOss4Source * osssrc, GstOss4SourceClass * g_class)
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{
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const gchar *device;
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device = g_getenv ("AUDIODEV");
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if (device == NULL)
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device = DEFAULT_DEVICE;
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osssrc->fd = -1;
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osssrc->device = g_strdup (device);
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osssrc->device_name = g_strdup (DEFAULT_DEVICE_NAME);
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osssrc->device_name = NULL;
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}
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static void
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gst_oss4_source_finalize (GstOss4Source * oss)
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{
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g_free (oss->device);
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oss->device = NULL;
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g_list_free (oss->property_probe_list);
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oss->property_probe_list = NULL;
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G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (oss));
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}
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static void
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gst_oss4_source_dispose (GObject * object)
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{
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GstOss4Source *oss = GST_OSS4_SOURCE (object);
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if (oss->probed_caps) {
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gst_caps_unref (oss->probed_caps);
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oss->probed_caps = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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gst_oss4_source_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstOss4Source *oss;
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oss = GST_OSS4_SOURCE (object);
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switch (prop_id) {
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case PROP_DEVICE:
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GST_OBJECT_LOCK (oss);
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if (oss->fd == -1) {
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g_free (oss->device);
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oss->device = g_value_dup_string (value);
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g_free (oss->device_name);
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oss->device_name = NULL;
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} else {
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g_warning ("%s: can't change \"device\" property while audio source "
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"is open", GST_OBJECT_NAME (oss));
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}
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GST_OBJECT_UNLOCK (oss);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_oss4_source_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstOss4Source *oss;
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oss = GST_OSS4_SOURCE (object);
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switch (prop_id) {
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case PROP_DEVICE:
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GST_OBJECT_LOCK (oss);
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g_value_set_string (value, oss->device);
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GST_OBJECT_UNLOCK (oss);
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break;
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case PROP_DEVICE_NAME:
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GST_OBJECT_LOCK (oss);
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/* If device is set, try to retrieve the name even if we're not open */
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if (oss->fd == -1 && oss->device != NULL) {
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if (gst_oss4_source_open (GST_AUDIO_SRC (oss), TRUE)) {
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g_value_set_string (value, oss->device_name);
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gst_oss4_source_close (GST_AUDIO_SRC (oss));
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} else {
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gchar *name = NULL;
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gst_oss4_property_probe_find_device_name_nofd (GST_OBJECT (oss),
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oss->device, &name);
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g_value_set_string (value, name);
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g_free (name);
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}
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} else {
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g_value_set_string (value, oss->device_name);
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}
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GST_OBJECT_UNLOCK (oss);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstCaps *
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gst_oss4_source_getcaps (GstBaseSrc * bsrc)
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{
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GstOss4Source *oss;
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GstCaps *caps;
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oss = GST_OSS4_SOURCE (bsrc);
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if (oss->fd == -1) {
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caps = gst_caps_copy (gst_oss4_audio_get_template_caps ());
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} else if (oss->probed_caps) {
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caps = gst_caps_copy (oss->probed_caps);
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} else {
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caps = gst_oss4_audio_probe_caps (GST_OBJECT (oss), oss->fd);
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if (caps != NULL && !gst_caps_is_empty (caps)) {
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oss->probed_caps = gst_caps_copy (caps);
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}
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}
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return caps;
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}
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/* note: we must not take the object lock here unless we fix up get_property */
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static gboolean
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gst_oss4_source_open (GstAudioSrc * asrc, gboolean silent_errors)
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{
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GstOss4Source *oss;
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gchar *device;
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int mode;
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oss = GST_OSS4_SOURCE (asrc);
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if (oss->device)
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device = g_strdup (oss->device);
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else
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device = gst_oss4_audio_find_device (GST_OBJECT_CAST (oss));
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/* desperate times, desperate measures */
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if (device == NULL)
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device = g_strdup ("/dev/dsp0");
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GST_INFO_OBJECT (oss, "Trying to open OSS4 device '%s'", device);
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/* we open in non-blocking mode even if we don't really want to do writes
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* non-blocking because we can't be sure that this is really a genuine
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* OSS4 device with well-behaved drivers etc. We really don't want to
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* hang forever under any circumstances. */
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oss->fd = open (device, O_RDONLY | O_NONBLOCK, 0);
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if (oss->fd == -1) {
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switch (errno) {
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case EBUSY:
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goto busy;
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case EACCES:
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goto no_permission;
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default:
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goto open_failed;
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}
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}
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GST_INFO_OBJECT (oss, "Opened device");
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/* Make sure it's OSS4. If it's old OSS, let osssink handle it */
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if (!gst_oss4_audio_check_version (GST_OBJECT_CAST (oss), oss->fd))
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goto legacy_oss;
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/* now remove the non-blocking flag. */
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mode = fcntl (oss->fd, F_GETFL);
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mode &= ~O_NONBLOCK;
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if (fcntl (oss->fd, F_SETFL, mode) < 0) {
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/* some drivers do no support unsetting the non-blocking flag, try to
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* close/open the device then. This is racy but we error out properly. */
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GST_WARNING_OBJECT (oss, "failed to unset O_NONBLOCK (buggy driver?), "
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"will try to re-open device now");
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gst_oss4_source_close (asrc);
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if ((oss->fd = open (device, O_RDONLY, 0)) == -1)
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goto non_block;
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}
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oss->open_device = device;
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/* not using ENGINEINFO here because it sometimes returns a different and
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* less useful name than AUDIOINFO for the same device */
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if (!gst_oss4_property_probe_find_device_name (GST_OBJECT (oss), oss->fd,
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oss->open_device, &oss->device_name)) {
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oss->device_name = NULL;
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}
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return TRUE;
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/* ERRORS */
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busy:
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{
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if (!silent_errors) {
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GST_ELEMENT_ERROR (oss, RESOURCE, BUSY,
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(_("Could not open audio device for playback. "
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"Device is being used by another application.")), (NULL));
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}
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g_free (device);
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return FALSE;
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}
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no_permission:
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{
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if (!silent_errors) {
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GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
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(_("Could not open audio device for playback. "
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"You don't have permission to open the device.")),
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GST_ERROR_SYSTEM);
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}
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g_free (device);
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return FALSE;
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}
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open_failed:
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{
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if (!silent_errors) {
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GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
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(_("Could not open audio device for playback.")), GST_ERROR_SYSTEM);
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}
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g_free (device);
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return FALSE;
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}
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legacy_oss:
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{
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gst_oss4_source_close (asrc);
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if (!silent_errors) {
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GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
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(_("Could not open audio device for playback. "
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"This version of the Open Sound System is not supported by this "
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"element.")), ("Try the 'osssink' element instead"));
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}
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g_free (device);
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return FALSE;
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}
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non_block:
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{
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if (!silent_errors) {
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GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),
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("Unable to set device %s into non-blocking mode: %s",
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oss->device, g_strerror (errno)));
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}
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g_free (device);
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return FALSE;
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}
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}
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static gboolean
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gst_oss4_source_open_func (GstAudioSrc * asrc)
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{
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return gst_oss4_source_open (asrc, FALSE);
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}
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static void
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gst_oss4_source_free_mixer_tracks (GstOss4Source * oss)
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{
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g_list_foreach (oss->tracks, (GFunc) g_object_unref, NULL);
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g_list_free (oss->tracks);
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oss->tracks = NULL;
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}
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static gboolean
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gst_oss4_source_close (GstAudioSrc * asrc)
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{
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GstOss4Source *oss;
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oss = GST_OSS4_SOURCE (asrc);
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if (oss->fd != -1) {
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GST_DEBUG_OBJECT (oss, "closing device");
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close (oss->fd);
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oss->fd = -1;
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}
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oss->bytes_per_sample = 0;
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gst_caps_replace (&oss->probed_caps, NULL);
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g_free (oss->open_device);
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oss->open_device = NULL;
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g_free (oss->device_name);
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oss->device_name = NULL;
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gst_oss4_source_free_mixer_tracks (oss);
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return TRUE;
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}
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static gboolean
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gst_oss4_source_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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{
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GstOss4Source *oss;
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oss = GST_OSS4_SOURCE (asrc);
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if (!gst_oss4_audio_set_format (GST_OBJECT_CAST (oss), oss->fd, spec)) {
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GST_WARNING_OBJECT (oss, "Couldn't set requested format %" GST_PTR_FORMAT,
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spec->caps);
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return FALSE;
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}
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oss->bytes_per_sample = spec->bytes_per_sample;
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return TRUE;
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}
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static gboolean
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gst_oss4_source_unprepare (GstAudioSrc * asrc)
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{
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/* could do a SNDCTL_DSP_HALT, but the OSS manual recommends a close/open,
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* since HALT won't properly reset some devices, apparently */
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if (!gst_oss4_source_close (asrc))
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|
goto couldnt_close;
|
|
|
|
if (!gst_oss4_source_open_func (asrc))
|
|
goto couldnt_reopen;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
couldnt_close:
|
|
{
|
|
GST_DEBUG_OBJECT (asrc, "Couldn't close the audio device");
|
|
return FALSE;
|
|
}
|
|
couldnt_reopen:
|
|
{
|
|
GST_DEBUG_OBJECT (asrc, "Couldn't reopen the audio device");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static guint
|
|
gst_oss4_source_read (GstAudioSrc * asrc, gpointer data, guint length)
|
|
{
|
|
GstOss4Source *oss;
|
|
int n;
|
|
|
|
oss = GST_OSS4_SOURCE_CAST (asrc);
|
|
|
|
n = read (oss->fd, data, length);
|
|
GST_LOG_OBJECT (asrc, "%u bytes, %u samples", n, n / oss->bytes_per_sample);
|
|
|
|
if (G_UNLIKELY (n < 0)) {
|
|
switch (errno) {
|
|
case ENOTSUP:
|
|
case EACCES:{
|
|
/* This is the most likely cause, I think */
|
|
GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
|
|
(_("Recording is not supported by this audio device.")),
|
|
("read: %s (device: %s) (maybe this is an output-only device?)",
|
|
g_strerror (errno), oss->open_device));
|
|
break;
|
|
}
|
|
default:{
|
|
GST_ELEMENT_ERROR (asrc, RESOURCE, READ,
|
|
(_("Error recording from audio device.")),
|
|
("read: %s (device: %s)", g_strerror (errno), oss->open_device));
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
return (guint) n;
|
|
}
|
|
|
|
static guint
|
|
gst_oss4_source_delay (GstAudioSrc * asrc)
|
|
{
|
|
audio_buf_info info = { 0, };
|
|
GstOss4Source *oss;
|
|
guint delay;
|
|
|
|
oss = GST_OSS4_SOURCE_CAST (asrc);
|
|
|
|
if (ioctl (oss->fd, SNDCTL_DSP_GETISPACE, &info) == -1) {
|
|
GST_LOG_OBJECT (oss, "GETISPACE failed: %s", g_strerror (errno));
|
|
return 0;
|
|
}
|
|
|
|
delay = (info.fragstotal * info.fragsize) - info.bytes;
|
|
GST_LOG_OBJECT (oss, "fragstotal:%d, fragsize:%d, bytes:%d, delay:%d",
|
|
info.fragstotal, info.fragsize, info.bytes, delay);
|
|
return delay;
|
|
}
|
|
|
|
static void
|
|
gst_oss4_source_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_HALT or similar */
|
|
}
|
|
|
|
/* GstMixer interface, which we abuse here for input selection, because we
|
|
* don't have a proper interface for that and because that's what
|
|
* gnome-sound-recorder does. */
|
|
|
|
/* GstMixerTrack is a plain GObject, so let's just use the GLib macro here */
|
|
G_DEFINE_TYPE (GstOss4SourceInput, gst_oss4_source_input, GST_TYPE_MIXER_TRACK);
|
|
|
|
static void
|
|
gst_oss4_source_input_class_init (GstOss4SourceInputClass * klass)
|
|
{
|
|
/* nothing to do here */
|
|
}
|
|
|
|
static void
|
|
gst_oss4_source_input_init (GstOss4SourceInput * i)
|
|
{
|
|
/* nothing to do here */
|
|
}
|
|
|
|
#if 0
|
|
|
|
static void
|
|
gst_ossmixer_ensure_track_list (GstOssMixer * mixer)
|
|
{
|
|
gint i, master = -1;
|
|
|
|
g_return_if_fail (mixer->fd != -1);
|
|
|
|
if (mixer->tracklist)
|
|
return;
|
|
|
|
/* find master volume */
|
|
if (mixer->devmask & SOUND_MASK_VOLUME)
|
|
master = SOUND_MIXER_VOLUME;
|
|
else if (mixer->devmask & SOUND_MASK_PCM)
|
|
master = SOUND_MIXER_PCM;
|
|
else if (mixer->devmask & SOUND_MASK_SPEAKER)
|
|
master = SOUND_MIXER_SPEAKER; /* doubtful... */
|
|
/* else: no master, so we won't set any */
|
|
|
|
/* build track list */
|
|
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
|
|
if (mixer->devmask & (1 << i)) {
|
|
GstMixerTrack *track;
|
|
gboolean input = FALSE, stereo = FALSE, record = FALSE;
|
|
|
|
/* track exists, make up capabilities */
|
|
if (MASK_BIT_IS_SET (mixer->stereomask, i))
|
|
stereo = TRUE;
|
|
if (MASK_BIT_IS_SET (mixer->recmask, i))
|
|
input = TRUE;
|
|
if (MASK_BIT_IS_SET (mixer->recdevs, i))
|
|
record = TRUE;
|
|
|
|
/* do we want mixer in our list? */
|
|
if (!((mixer->dir & GST_OSS_MIXER_CAPTURE && input == TRUE) ||
|
|
(mixer->dir & GST_OSS_MIXER_PLAYBACK && i != SOUND_MIXER_PCM)))
|
|
/* the PLAYBACK case seems hacky, but that's how 0.8 had it */
|
|
continue;
|
|
|
|
/* add track to list */
|
|
track = gst_ossmixer_track_new (mixer->fd, i, stereo ? 2 : 1,
|
|
(record ? GST_MIXER_TRACK_RECORD : 0) |
|
|
(input ? GST_MIXER_TRACK_INPUT :
|
|
GST_MIXER_TRACK_OUTPUT) |
|
|
((master != i) ? 0 : GST_MIXER_TRACK_MASTER));
|
|
mixer->tracklist = g_list_append (mixer->tracklist, track);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* unused with G_DISABLE_* */
|
|
static G_GNUC_UNUSED gboolean
|
|
gst_ossmixer_contains_track (GstOssMixer * mixer, GstOssMixerTrack * osstrack)
|
|
{
|
|
const GList *item;
|
|
|
|
for (item = mixer->tracklist; item != NULL; item = item->next)
|
|
if (item->data == osstrack)
|
|
return TRUE;
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
const GList *
|
|
gst_ossmixer_list_tracks (GstOssMixer * mixer)
|
|
{
|
|
gst_ossmixer_ensure_track_list (mixer);
|
|
|
|
return (const GList *) mixer->tracklist;
|
|
}
|
|
|
|
void
|
|
gst_ossmixer_get_volume (GstOssMixer * mixer,
|
|
GstMixerTrack * track, gint * volumes)
|
|
{
|
|
gint volume;
|
|
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
|
|
|
|
g_return_if_fail (mixer->fd != -1);
|
|
g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
|
|
|
|
if (track->flags & GST_MIXER_TRACK_MUTE) {
|
|
volumes[0] = osstrack->lvol;
|
|
if (track->num_channels == 2) {
|
|
volumes[1] = osstrack->rvol;
|
|
}
|
|
} else {
|
|
/* get */
|
|
if (ioctl (mixer->fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
|
|
g_warning ("Error getting recording device (%d) volume: %s",
|
|
osstrack->track_num, g_strerror (errno));
|
|
volume = 0;
|
|
}
|
|
|
|
osstrack->lvol = volumes[0] = (volume & 0xff);
|
|
if (track->num_channels == 2) {
|
|
osstrack->rvol = volumes[1] = ((volume >> 8) & 0xff);
|
|
}
|
|
}
|
|
}
|
|
|
|
void
|
|
gst_ossmixer_set_mute (GstOssMixer * mixer, GstMixerTrack * track,
|
|
gboolean mute)
|
|
{
|
|
int volume;
|
|
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
|
|
|
|
g_return_if_fail (mixer->fd != -1);
|
|
g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
|
|
|
|
if (mute) {
|
|
volume = 0;
|
|
} else {
|
|
volume = (osstrack->lvol & 0xff);
|
|
if (MASK_BIT_IS_SET (mixer->stereomask, osstrack->track_num)) {
|
|
volume |= ((osstrack->rvol & 0xff) << 8);
|
|
}
|
|
}
|
|
|
|
if (ioctl (mixer->fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
|
|
g_warning ("Error setting mixer recording device volume (0x%x): %s",
|
|
volume, g_strerror (errno));
|
|
return;
|
|
}
|
|
|
|
if (mute) {
|
|
track->flags |= GST_MIXER_TRACK_MUTE;
|
|
} else {
|
|
track->flags &= ~GST_MIXER_TRACK_MUTE;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
static gint
|
|
gst_oss4_source_mixer_get_current_input (GstOss4Source * oss)
|
|
{
|
|
int cur = -1;
|
|
|
|
if (ioctl (oss->fd, SNDCTL_DSP_GET_RECSRC, &cur) == -1 || cur < 0)
|
|
return -1;
|
|
|
|
return cur;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_oss4_source_mixer_update_record_flags (GstOss4Source * oss, gint cur_route)
|
|
{
|
|
const gchar *cur_name = "";
|
|
GList *t;
|
|
|
|
for (t = oss->tracks; t != NULL; t = t->next) {
|
|
GstMixerTrack *track = t->data;
|
|
|
|
if (GST_OSS4_SOURCE_INPUT (track)->route == cur_route) {
|
|
if (!GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) {
|
|
track->flags |= GST_MIXER_TRACK_RECORD;
|
|
/* no point in sending a mixer-record-changes message here */
|
|
}
|
|
cur_name = track->label;
|
|
} else {
|
|
if (GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) {
|
|
track->flags &= ~GST_MIXER_TRACK_RECORD;
|
|
/* no point in sending a mixer-record-changes message here */
|
|
}
|
|
}
|
|
}
|
|
|
|
return cur_name;
|
|
}
|
|
|
|
static const GList *
|
|
gst_oss4_source_mixer_list_tracks (GstMixer * mixer)
|
|
{
|
|
oss_mixer_enuminfo names = { 0, };
|
|
GstOss4Source *oss;
|
|
const gchar *cur_name;
|
|
GList *tracks = NULL;
|
|
gint i, cur;
|
|
|
|
g_return_val_if_fail (mixer != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_OSS4_SOURCE (mixer), NULL);
|
|
g_return_val_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer), NULL);
|
|
|
|
oss = GST_OSS4_SOURCE (mixer);
|
|
|
|
if (oss->tracks != NULL && oss->tracks_static)
|
|
goto done;
|
|
|
|
if (ioctl (oss->fd, SNDCTL_DSP_GET_RECSRC_NAMES, &names) == -1)
|
|
goto get_recsrc_names_error;
|
|
|
|
oss->tracks_static = (names.version == 0);
|
|
|
|
GST_INFO_OBJECT (oss, "%d inputs (list is static: %s):", names.nvalues,
|
|
(oss->tracks_static) ? "yes" : "no");
|
|
|
|
for (i = 0; i < MIN (names.nvalues, OSS_ENUM_MAXVALUE + 1); ++i) {
|
|
GstMixerTrack *track;
|
|
|
|
track = g_object_new (GST_TYPE_OSS4_SOURCE_INPUT, NULL);
|
|
track->label = g_strdup (&names.strings[names.strindex[i]]);
|
|
track->flags = GST_MIXER_TRACK_INPUT;
|
|
track->num_channels = 2;
|
|
track->min_volume = 0;
|
|
track->max_volume = 100;
|
|
GST_OSS4_SOURCE_INPUT (track)->route = i;
|
|
|
|
GST_INFO_OBJECT (oss, " [%d] %s", i, track->label);
|
|
tracks = g_list_append (tracks, track);
|
|
}
|
|
|
|
gst_oss4_source_free_mixer_tracks (oss);
|
|
oss->tracks = tracks;
|
|
|
|
done:
|
|
|
|
/* update RECORD flags */
|
|
cur = gst_oss4_source_mixer_get_current_input (oss);
|
|
cur_name = gst_oss4_source_mixer_update_record_flags (oss, cur);
|
|
GST_DEBUG_OBJECT (oss, "current input route: %d (%s)", cur, cur_name);
|
|
|
|
return (const GList *) oss->tracks;
|
|
|
|
/* ERRORS */
|
|
get_recsrc_names_error:
|
|
{
|
|
GST_WARNING_OBJECT (oss, "GET_RECSRC_NAMES failed: %s", g_strerror (errno));
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_oss4_source_mixer_set_volume (GstMixer * mixer, GstMixerTrack * track,
|
|
gint * volumes)
|
|
{
|
|
GstOss4Source *oss;
|
|
int new_vol, cur;
|
|
|
|
g_return_if_fail (mixer != NULL);
|
|
g_return_if_fail (track != NULL);
|
|
g_return_if_fail (GST_IS_MIXER_TRACK (track));
|
|
g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
|
|
g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
|
|
|
|
oss = GST_OSS4_SOURCE (mixer);
|
|
|
|
cur = gst_oss4_source_mixer_get_current_input (oss);
|
|
if (cur != GST_OSS4_SOURCE_INPUT (track)->route) {
|
|
GST_DEBUG_OBJECT (oss, "track not selected input route, ignoring request");
|
|
return;
|
|
}
|
|
|
|
new_vol = (volumes[1] << 8) | volumes[0];
|
|
if (ioctl (oss->fd, SNDCTL_DSP_SETRECVOL, &new_vol) == -1) {
|
|
GST_WARNING_OBJECT (oss, "SETRECVOL failed: %s", g_strerror (errno));
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_oss4_source_mixer_get_volume (GstMixer * mixer, GstMixerTrack * track,
|
|
gint * volumes)
|
|
{
|
|
GstOss4Source *oss;
|
|
int cur;
|
|
|
|
g_return_if_fail (mixer != NULL);
|
|
g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
|
|
g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
|
|
|
|
oss = GST_OSS4_SOURCE (mixer);
|
|
|
|
cur = gst_oss4_source_mixer_get_current_input (oss);
|
|
if (cur != GST_OSS4_SOURCE_INPUT (track)->route) {
|
|
volumes[0] = 0;
|
|
volumes[1] = 0;
|
|
} else {
|
|
int vol = -1;
|
|
|
|
if (ioctl (oss->fd, SNDCTL_DSP_GETRECVOL, &vol) == -1 || vol < 0) {
|
|
GST_WARNING_OBJECT (oss, "GETRECVOL failed: %s", g_strerror (errno));
|
|
volumes[0] = 100;
|
|
volumes[1] = 100;
|
|
} else {
|
|
volumes[0] = MIN (100, vol & 0xff);
|
|
volumes[1] = MIN (100, (vol >> 8) & 0xff);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_oss4_source_mixer_set_record (GstMixer * mixer, GstMixerTrack * track,
|
|
gboolean record)
|
|
{
|
|
GstOss4Source *oss;
|
|
const gchar *cur_name;
|
|
gint cur;
|
|
|
|
g_return_if_fail (mixer != NULL);
|
|
g_return_if_fail (track != NULL);
|
|
g_return_if_fail (GST_IS_MIXER_TRACK (track));
|
|
g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
|
|
g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
|
|
|
|
oss = GST_OSS4_SOURCE (mixer);
|
|
|
|
cur = gst_oss4_source_mixer_get_current_input (oss);
|
|
|
|
/* stop recording for an input that's not selected anyway => nothing to do */
|
|
if (!record && cur != GST_OSS4_SOURCE_INPUT (track)->route)
|
|
goto done;
|
|
|
|
/* select recording for an input that's already selected => nothing to do
|
|
* (or should we mess with the recording volume in this case maybe?) */
|
|
if (record && cur == GST_OSS4_SOURCE_INPUT (track)->route)
|
|
goto done;
|
|
|
|
/* make current input stop recording: we can't really make an input stop
|
|
* recording, we can only select an input FOR recording, so we'll just ignore
|
|
* all requests to stop for now */
|
|
if (!record) {
|
|
GST_WARNING_OBJECT (oss, "Can't un-select an input as such, only switch "
|
|
"to a different input source");
|
|
/* FIXME: set recording volume to 0 maybe? */
|
|
} else {
|
|
int new_route = GST_OSS4_SOURCE_INPUT (track)->route;
|
|
|
|
/* select this input for recording */
|
|
|
|
if (ioctl (oss->fd, SNDCTL_DSP_SET_RECSRC, &new_route) == -1) {
|
|
GST_WARNING_OBJECT (oss, "Could not select input %d for recording: %s",
|
|
new_route, g_strerror (errno));
|
|
} else {
|
|
cur = new_route;
|
|
}
|
|
}
|
|
|
|
done:
|
|
|
|
cur_name = gst_oss4_source_mixer_update_record_flags (oss, cur);
|
|
GST_DEBUG_OBJECT (oss, "active input route: %d (%s)", cur, cur_name);
|
|
}
|
|
|
|
static void
|
|
gst_oss4_source_mixer_set_mute (GstMixer * mixer, GstMixerTrack * track,
|
|
gboolean mute)
|
|
{
|
|
GstOss4Source *oss;
|
|
|
|
g_return_if_fail (mixer != NULL);
|
|
g_return_if_fail (track != NULL);
|
|
g_return_if_fail (GST_IS_MIXER_TRACK (track));
|
|
g_return_if_fail (GST_IS_OSS4_SOURCE (mixer));
|
|
g_return_if_fail (GST_OSS4_SOURCE_IS_OPEN (mixer));
|
|
|
|
oss = GST_OSS4_SOURCE (mixer);
|
|
|
|
/* FIXME: implement gst_oss4_source_mixer_set_mute() - what to do here? */
|
|
/* oss4_mixer_set_mute (mixer->mixer, track, mute); */
|
|
}
|
|
|
|
static void
|
|
gst_oss4_source_mixer_interface_init (GstMixerClass * klass)
|
|
{
|
|
GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE;
|
|
|
|
klass->list_tracks = gst_oss4_source_mixer_list_tracks;
|
|
klass->set_volume = gst_oss4_source_mixer_set_volume;
|
|
klass->get_volume = gst_oss4_source_mixer_get_volume;
|
|
klass->set_mute = gst_oss4_source_mixer_set_mute;
|
|
klass->set_record = gst_oss4_source_mixer_set_record;
|
|
}
|
|
|
|
/* Implement the horror that is GstImplementsInterface */
|
|
|
|
static gboolean
|
|
gst_oss4_source_mixer_supported (GstImplementsInterface * iface,
|
|
GType iface_type)
|
|
{
|
|
GstOss4Source *oss;
|
|
gboolean is_open;
|
|
|
|
g_return_val_if_fail (GST_IS_OSS4_SOURCE (iface), FALSE);
|
|
g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
|
|
|
|
oss = GST_OSS4_SOURCE (iface);
|
|
|
|
GST_OBJECT_LOCK (oss);
|
|
is_open = GST_OSS4_SOURCE_IS_OPEN (iface);
|
|
GST_OBJECT_UNLOCK (oss);
|
|
|
|
return is_open;
|
|
}
|
|
|
|
static void
|
|
gst_oss4_source_mixer_implements_interface_init (GstImplementsInterfaceClass *
|
|
klass)
|
|
{
|
|
klass->supported = gst_oss4_source_mixer_supported;
|
|
}
|
|
|
|
static void
|
|
gst_oss4_source_init_interfaces (GType type)
|
|
{
|
|
static const GInterfaceInfo implements_iface_info = {
|
|
(GInterfaceInitFunc) gst_oss4_source_mixer_implements_interface_init,
|
|
NULL,
|
|
NULL,
|
|
};
|
|
static const GInterfaceInfo mixer_iface_info = {
|
|
(GInterfaceInitFunc) gst_oss4_source_mixer_interface_init,
|
|
NULL,
|
|
NULL,
|
|
};
|
|
|
|
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
|
|
&implements_iface_info);
|
|
g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
|
|
|
|
gst_oss4_add_property_probe_interface (type);
|
|
}
|