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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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537 lines
14 KiB
C
537 lines
14 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000,2005 Wim Taymans <wim@fluendo.com>
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*
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* gstosssrc.c:
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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-osssrc
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*
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* This element lets you record sound using the Open Sound System (OSS).
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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 osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
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* ]| will record sound from your sound card using OSS 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 enabled etc.)
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* </refsect2>
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*/
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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/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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#ifdef HAVE_OSS_INCLUDE_IN_SYS
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# include <sys/soundcard.h>
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#else
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# ifdef HAVE_OSS_INCLUDE_IN_ROOT
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# include <soundcard.h>
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# else
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# ifdef HAVE_OSS_INCLUDE_IN_MACHINE
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# include <machine/soundcard.h>
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# else
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# error "What to include?"
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# endif /* HAVE_OSS_INCLUDE_IN_MACHINE */
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# endif /* HAVE_OSS_INCLUDE_IN_ROOT */
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#endif /* HAVE_OSS_INCLUDE_IN_SYS */
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#include "gstosssrc.h"
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#include "common.h"
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#include <gst/gst-i18n-plugin.h>
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GST_DEBUG_CATEGORY_EXTERN (oss_debug);
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#define GST_CAT_DEFAULT oss_debug
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#define DEFAULT_DEVICE "/dev/dsp"
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#define DEFAULT_DEVICE_NAME ""
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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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GST_BOILERPLATE_WITH_INTERFACE (GstOssSrc, gst_oss_src, GstAudioSrc,
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GST_TYPE_AUDIO_SRC, GstMixer, GST_TYPE_MIXER, gst_oss_src_mixer);
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GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssSrc, gst_oss_src_mixer);
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static void gst_oss_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_oss_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_oss_src_dispose (GObject * object);
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static void gst_oss_src_finalize (GstOssSrc * osssrc);
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static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc);
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static gboolean gst_oss_src_open (GstAudioSrc * asrc);
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static gboolean gst_oss_src_close (GstAudioSrc * asrc);
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static gboolean gst_oss_src_prepare (GstAudioSrc * asrc,
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GstRingBufferSpec * spec);
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static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc);
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static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length);
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static guint gst_oss_src_delay (GstAudioSrc * asrc);
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static void gst_oss_src_reset (GstAudioSrc * asrc);
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static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
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"audio/x-raw-int, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) 8, "
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"depth = (int) 8, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
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);
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static void
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gst_oss_src_dispose (GObject * object)
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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_oss_src_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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gst_element_class_set_details_simple (element_class, "Audio Source (OSS)",
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"Source/Audio",
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"Capture from a sound card via OSS",
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"Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&osssrc_src_factory));
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}
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static void
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gst_oss_src_class_init (GstOssSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseSrcClass *gstbasesrc_class;
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GstAudioSrcClass *gstaudiosrc_class;
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gobject_class = (GObjectClass *) klass;
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gstbasesrc_class = (GstBaseSrcClass *) klass;
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gstaudiosrc_class = (GstAudioSrcClass *) klass;
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gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss_src_dispose);
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gobject_class->finalize =
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(GObjectFinalizeFunc) GST_DEBUG_FUNCPTR (gst_oss_src_finalize);
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gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_oss_src_get_property);
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_oss_src_set_property);
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_src_getcaps);
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gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss_src_open);
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gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_src_prepare);
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gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_src_unprepare);
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gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss_src_close);
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gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss_src_read);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss_src_delay);
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gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss_src_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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"OSS device (usually /dev/dspN)", 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_oss_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstOssSrc *src;
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src = GST_OSS_SRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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if (src->device)
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g_free (src->device);
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src->device = g_value_dup_string (value);
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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_oss_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstOssSrc *src;
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src = GST_OSS_SRC (object);
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switch (prop_id) {
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case PROP_DEVICE:
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g_value_set_string (value, src->device);
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break;
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case PROP_DEVICE_NAME:
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g_value_set_string (value, src->device_name);
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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_oss_src_init (GstOssSrc * osssrc, GstOssSrcClass * g_class)
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{
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const gchar *device;
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GST_DEBUG ("initializing osssrc");
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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->probed_caps = NULL;
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}
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static void
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gst_oss_src_finalize (GstOssSrc * osssrc)
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{
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g_free (osssrc->device);
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g_free (osssrc->device_name);
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G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (osssrc));
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}
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static GstCaps *
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gst_oss_src_getcaps (GstBaseSrc * bsrc)
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{
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GstOssSrc *osssrc;
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GstCaps *caps;
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osssrc = GST_OSS_SRC (bsrc);
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if (osssrc->fd == -1) {
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GST_DEBUG_OBJECT (osssrc, "device not open, using template caps");
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return NULL; /* base class will get template caps for us */
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}
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if (osssrc->probed_caps) {
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GST_LOG_OBJECT (osssrc, "Returning cached caps");
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return gst_caps_ref (osssrc->probed_caps);
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}
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caps = gst_oss_helper_probe_caps (osssrc->fd);
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if (caps) {
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osssrc->probed_caps = gst_caps_ref (caps);
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}
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GST_INFO_OBJECT (osssrc, "returning caps %" GST_PTR_FORMAT, caps);
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return caps;
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}
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static gint
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ilog2 (gint x)
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{
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/* well... hacker's delight explains... */
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x = x | (x >> 1);
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x = x | (x >> 2);
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x = x | (x >> 4);
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x = x | (x >> 8);
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x = x | (x >> 16);
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x = x - ((x >> 1) & 0x55555555);
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x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
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x = (x + (x >> 4)) & 0x0f0f0f0f;
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x = x + (x >> 8);
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x = x + (x >> 16);
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return (x & 0x0000003f) - 1;
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}
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static gint
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gst_oss_src_get_format (GstBufferFormat fmt)
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{
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gint result;
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switch (fmt) {
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case GST_MU_LAW:
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result = AFMT_MU_LAW;
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break;
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case GST_A_LAW:
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result = AFMT_A_LAW;
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break;
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case GST_IMA_ADPCM:
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result = AFMT_IMA_ADPCM;
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break;
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case GST_U8:
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result = AFMT_U8;
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break;
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case GST_S16_LE:
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result = AFMT_S16_LE;
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break;
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case GST_S16_BE:
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result = AFMT_S16_BE;
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break;
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case GST_S8:
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result = AFMT_S8;
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break;
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case GST_U16_LE:
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result = AFMT_U16_LE;
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break;
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case GST_U16_BE:
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result = AFMT_U16_BE;
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break;
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case GST_MPEG:
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result = AFMT_MPEG;
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break;
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default:
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result = 0;
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break;
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}
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return result;
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}
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static gboolean
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gst_oss_src_open (GstAudioSrc * asrc)
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{
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GstOssSrc *oss;
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int mode;
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oss = GST_OSS_SRC (asrc);
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mode = O_RDONLY;
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mode |= O_NONBLOCK;
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oss->fd = open (oss->device, mode, 0);
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if (oss->fd == -1) {
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switch (errno) {
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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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if (!oss->mixer) {
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oss->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_CAPTURE);
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if (oss->mixer) {
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g_free (oss->device_name);
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oss->device_name = g_strdup (oss->mixer->cardname);
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}
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}
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return TRUE;
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no_permission:
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{
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GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
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(_("Could not open audio device for recording. "
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"You don't have permission to open the device.")),
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GST_ERROR_SYSTEM);
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return FALSE;
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}
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open_failed:
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{
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GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
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(_("Could not open audio device for recording.")),
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("Unable to open device %s for recording: %s",
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oss->device, g_strerror (errno)));
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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_oss_src_close (GstAudioSrc * asrc)
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{
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GstOssSrc *oss;
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oss = GST_OSS_SRC (asrc);
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close (oss->fd);
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if (oss->mixer) {
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gst_ossmixer_free (oss->mixer);
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oss->mixer = NULL;
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}
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gst_caps_replace (&oss->probed_caps, NULL);
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return TRUE;
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}
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static gboolean
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gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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{
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GstOssSrc *oss;
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struct audio_buf_info info;
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int mode;
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int fmt, tmp;
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oss = GST_OSS_SRC (asrc);
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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) == -1)
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goto non_block;
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fmt = gst_oss_src_get_format (spec->format);
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if (fmt == 0)
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goto wrong_format;
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tmp = ilog2 (spec->segsize);
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tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
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GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",
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spec->segsize, spec->segtotal, tmp);
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SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT");
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SET_PARAM (oss, SNDCTL_DSP_RESET, 0, "RESET");
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SET_PARAM (oss, SNDCTL_DSP_SETFMT, fmt, "SETFMT");
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if (spec->channels == 2)
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SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");
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SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels, "CHANNELS");
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SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate, "SPEED");
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GET_PARAM (oss, SNDCTL_DSP_GETISPACE, &info, "GETISPACE");
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spec->segsize = info.fragsize;
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spec->segtotal = info.fragstotal;
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if (spec->width != 16 && spec->width != 8)
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goto dodgy_width;
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spec->bytes_per_sample = (spec->width / 8) * spec->channels;
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oss->bytes_per_sample = (spec->width / 8) * spec->channels;
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memset (spec->silence_sample, 0, spec->bytes_per_sample);
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GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",
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spec->segsize, spec->segtotal, tmp);
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return TRUE;
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non_block:
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{
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GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
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("Unable to set device %s in non blocking mode: %s",
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oss->device, g_strerror (errno)), (NULL));
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return FALSE;
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}
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wrong_format:
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{
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GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
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("Unable to get format %d", spec->format), (NULL));
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return FALSE;
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}
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dodgy_width:
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{
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GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ,
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("Unexpected width %d", spec->width), (NULL));
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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_oss_src_unprepare (GstAudioSrc * asrc)
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{
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/* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
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if (!gst_oss_src_close (asrc))
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goto couldnt_close;
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if (!gst_oss_src_open (asrc))
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goto couldnt_reopen;
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return TRUE;
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couldnt_close:
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{
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GST_DEBUG_OBJECT (asrc, "Could not close the audio device");
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return FALSE;
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}
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couldnt_reopen:
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{
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GST_DEBUG_OBJECT (asrc, "Could not reopen the audio device");
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return FALSE;
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}
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}
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static guint
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gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length)
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{
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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 */
|
|
}
|