/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2000,2005 Wim Taymans * * gstosssink.c: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #include #include "gstosssink.h" /* elementfactory information */ static GstElementDetails gst_oss_sink_details = GST_ELEMENT_DETAILS ("Audio Sink (OSS)", "Sink/Audio", "Output to a sound card via OSS", "Erik Walthinsen , " "Wim Taymans "); static void gst_oss_sink_base_init (gpointer g_class); static void gst_oss_sink_class_init (GstOssSinkClass * klass); static void gst_oss_sink_init (GstOssSink * osssink); static void gst_oss_sink_dispose (GObject * object); static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink); static gboolean gst_oss_sink_open (GstAudioSink * asink); static gboolean gst_oss_sink_close (GstAudioSink * asink); static gboolean gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec); static gboolean gst_oss_sink_unprepare (GstAudioSink * asink); static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length); static guint gst_oss_sink_delay (GstAudioSink * asink); static void gst_oss_sink_reset (GstAudioSink * asink); /* OssSink signals and args */ enum { LAST_SIGNAL }; static GstStaticPadTemplate osssink_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, 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 GstElementClass *parent_class = NULL; /* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */ GType gst_oss_sink_get_type (void) { static GType osssink_type = 0; if (!osssink_type) { static const GTypeInfo osssink_info = { sizeof (GstOssSinkClass), gst_oss_sink_base_init, NULL, (GClassInitFunc) gst_oss_sink_class_init, NULL, NULL, sizeof (GstOssSink), 0, (GInstanceInitFunc) gst_oss_sink_init, }; osssink_type = g_type_register_static (GST_TYPE_AUDIO_SINK, "GstOssSink", &osssink_info, 0); } return osssink_type; } static void gst_oss_sink_dispose (GObject * object) { G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_oss_sink_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_set_details (element_class, &gst_oss_sink_details); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&osssink_sink_factory)); } static void gst_oss_sink_class_init (GstOssSinkClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseSinkClass *gstbasesink_class; GstBaseAudioSinkClass *gstbaseaudiosink_class; GstAudioSinkClass *gstaudiosink_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasesink_class = (GstBaseSinkClass *) klass; gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; gstaudiosink_class = (GstAudioSinkClass *) klass; parent_class = g_type_class_ref (GST_TYPE_BASE_AUDIO_SINK); gobject_class->dispose = gst_oss_sink_dispose; gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps); gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open); gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close); gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare); gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare); gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write); gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay); gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset); } static void gst_oss_sink_init (GstOssSink * osssink) { GST_DEBUG ("initializing osssink"); osssink->element = g_object_new (GST_TYPE_OSSELEMENT, NULL); } static GstCaps * gst_oss_sink_getcaps (GstBaseSink * bsink) { GstOssSink *osssink; GstOssElement *element; GstCaps *caps; osssink = GST_OSSSINK (bsink); element = osssink->element; gst_osselement_probe_caps (element); if (element->probed_caps == NULL) { caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink))); } else { caps = gst_caps_ref (element->probed_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; } #define SET_PARAM(_oss, _name, _val) \ G_STMT_START { \ int _tmp = _val; \ if (ioctl(_oss->fd, _name, &_tmp) == -1) { \ perror(G_STRINGIFY (_name)); \ return FALSE; \ } \ GST_DEBUG(G_STRINGIFY (name) " %d", _tmp); \ } G_STMT_END #define GET_PARAM(oss, name, val) \ G_STMT_START { \ if (ioctl(oss->fd, name, val) == -1) { \ perror(G_STRINGIFY (name)); \ return FALSE; \ } \ } G_STMT_END static gint gst_oss_sink_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_sink_open (GstAudioSink * asink) { GstOssSink *oss; int mode; oss = GST_OSSSINK (asink); mode = O_WRONLY; mode |= O_NONBLOCK; oss->fd = open ("/dev/dsp", mode, 0); if (oss->fd == -1) { perror ("/dev/dsp"); return FALSE; } return TRUE; } static gboolean gst_oss_sink_close (GstAudioSink * asink) { close (GST_OSSSINK (asink)->fd); return TRUE; } static gboolean gst_oss_sink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec) { GstOssSink *oss; struct audio_buf_info info; int mode; int tmp; oss = GST_OSSSINK (asink); mode = fcntl (oss->fd, F_GETFL); mode &= ~O_NONBLOCK; if (fcntl (oss->fd, F_SETFL, mode) == -1) { perror ("/dev/dsp"); return FALSE; } tmp = gst_oss_sink_get_format (spec->format); if (tmp == 0) goto wrong_format; SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp); if (spec->channels == 2) SET_PARAM (oss, SNDCTL_DSP_STEREO, 1); SET_PARAM (oss, SNDCTL_DSP_CHANNELS, spec->channels); SET_PARAM (oss, SNDCTL_DSP_SPEED, spec->rate); tmp = ilog2 (spec->segsize); tmp = ((spec->segtotal & 0x7fff) << 16) | tmp; GST_DEBUG ("set segsize: %d, segtotal: %d, value: %08x", spec->segsize, spec->segtotal, tmp); SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp); GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info); spec->segsize = info.fragsize; spec->segtotal = info.fragstotal; spec->bytes_per_sample = 4; oss->bytes_per_sample = 4; memset (spec->silence_sample, 0, spec->bytes_per_sample); GST_DEBUG ("got segsize: %d, segtotal: %d, value: %08x", spec->segsize, spec->segtotal, tmp); return TRUE; wrong_format: { GST_DEBUG ("wrong format %d\n", spec->format); return FALSE; } } static gboolean gst_oss_sink_unprepare (GstAudioSink * asink) { /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */ if (!gst_oss_sink_close (asink)) goto couldnt_close; if (!gst_oss_sink_open (asink)) goto couldnt_reopen; return TRUE; couldnt_close: { GST_DEBUG ("Could not close the audio device"); return FALSE; } couldnt_reopen: { GST_DEBUG ("Could not reopen the audio device"); return FALSE; } } static guint gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length) { return write (GST_OSSSINK (asink)->fd, data, length); } static guint gst_oss_sink_delay (GstAudioSink * asink) { GstOssSink *oss; gint delay = 0; gint ret; oss = GST_OSSSINK (asink); #ifdef SNDCTL_DSP_GETODELAY ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay); #else ret = -1; #endif if (ret < 0) { audio_buf_info info; ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info); delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes); } return delay / oss->bytes_per_sample; } static void gst_oss_sink_reset (GstAudioSink * asink) { GstOssSink *oss; //gint ret; oss = GST_OSSSINK (asink); /* deadlocks on my machine... */ //ret = ioctl (oss->fd, SNDCTL_DSP_RESET, 0); }