/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2000,2005 Wim Taymans * * gstosssrc.c: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:element-osssrc * * This element lets you record sound using the Open Sound System (OSS). * * * Example pipelines * |[ * gst-launch -v osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg * ]| will record sound from your sound card using OSS and encode it to an * Ogg/Vorbis file (this will only work if your mixer settings are right * and the right inputs enabled etc.) * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #ifdef HAVE_OSS_INCLUDE_IN_SYS # include #else # ifdef HAVE_OSS_INCLUDE_IN_ROOT # include # else # ifdef HAVE_OSS_INCLUDE_IN_MACHINE # include # else # error "What to include?" # endif /* HAVE_OSS_INCLUDE_IN_MACHINE */ # endif /* HAVE_OSS_INCLUDE_IN_ROOT */ #endif /* HAVE_OSS_INCLUDE_IN_SYS */ #include "gstosssrc.h" #include "common.h" #include GST_DEBUG_CATEGORY_EXTERN (oss_debug); #define GST_CAT_DEFAULT oss_debug #define DEFAULT_DEVICE "/dev/dsp" #define DEFAULT_DEVICE_NAME "" enum { PROP_0, PROP_DEVICE, PROP_DEVICE_NAME, }; #define gst_oss_src_parent_class parent_class G_DEFINE_TYPE (GstOssSrc, gst_oss_src, GST_TYPE_AUDIO_SRC); static void gst_oss_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_oss_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_oss_src_dispose (GObject * object); static void gst_oss_src_finalize (GstOssSrc * osssrc); static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter); static gboolean gst_oss_src_open (GstAudioSrc * asrc); static gboolean gst_oss_src_close (GstAudioSrc * asrc); static gboolean gst_oss_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec); static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc); static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length); static guint gst_oss_src_delay (GstAudioSrc * asrc); static void gst_oss_src_reset (GstAudioSrc * asrc); #define FORMATS "{" GST_AUDIO_NE(S16)","GST_AUDIO_NE(U16)", S8, U8 }" static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " FORMATS ", " "layout = (string) interleaved, " "rate = (int) [ 1, MAX ], " "channels = (int) 1; " "audio/x-raw, " "format = (string) " FORMATS ", " "layout = (string) interleaved, " "rate = (int) [ 1, MAX ], " "channels = (int) 2, " "channel-mask = (bitmask) 0x3") ); static void gst_oss_src_dispose (GObject * object) { G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_oss_src_class_init (GstOssSrcClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseSrcClass *gstbasesrc_class; GstAudioSrcClass *gstaudiosrc_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbasesrc_class = (GstBaseSrcClass *) klass; gstaudiosrc_class = (GstAudioSrcClass *) klass; gobject_class->dispose = gst_oss_src_dispose; gobject_class->finalize = (GObjectFinalizeFunc) gst_oss_src_finalize; gobject_class->get_property = gst_oss_src_get_property; gobject_class->set_property = gst_oss_src_set_property; gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_src_getcaps); gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss_src_open); gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_src_prepare); gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_src_unprepare); gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss_src_close); gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss_src_read); gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss_src_delay); gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss_src_reset); g_object_class_install_property (gobject_class, PROP_DEVICE, g_param_spec_string ("device", "Device", "OSS device (usually /dev/dspN)", DEFAULT_DEVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_DEVICE_NAME, g_param_spec_string ("device-name", "Device name", "Human-readable name of the sound device", DEFAULT_DEVICE_NAME, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); gst_element_class_set_static_metadata (gstelement_class, "Audio Source (OSS)", "Source/Audio", "Capture from a sound card via OSS", "Erik Walthinsen , " "Wim Taymans "); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&osssrc_src_factory)); } static void gst_oss_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstOssSrc *src; src = GST_OSS_SRC (object); switch (prop_id) { case PROP_DEVICE: if (src->device) g_free (src->device); src->device = g_value_dup_string (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_oss_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstOssSrc *src; src = GST_OSS_SRC (object); switch (prop_id) { case PROP_DEVICE: g_value_set_string (value, src->device); break; case PROP_DEVICE_NAME: g_value_set_string (value, src->device_name); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_oss_src_init (GstOssSrc * osssrc) { const gchar *device; GST_DEBUG ("initializing osssrc"); device = g_getenv ("AUDIODEV"); if (device == NULL) device = DEFAULT_DEVICE; osssrc->fd = -1; osssrc->device = g_strdup (device); osssrc->device_name = g_strdup (DEFAULT_DEVICE_NAME); osssrc->probed_caps = NULL; } static void gst_oss_src_finalize (GstOssSrc * osssrc) { g_free (osssrc->device); g_free (osssrc->device_name); G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (osssrc)); } static GstCaps * gst_oss_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter) { GstOssSrc *osssrc; GstCaps *caps; osssrc = GST_OSS_SRC (bsrc); if (osssrc->fd == -1) { GST_DEBUG_OBJECT (osssrc, "device not open, using template caps"); return NULL; /* base class will get template caps for us */ } if (osssrc->probed_caps) { GST_LOG_OBJECT (osssrc, "Returning cached caps"); return gst_caps_ref (osssrc->probed_caps); } caps = gst_oss_helper_probe_caps (osssrc->fd); if (caps) { osssrc->probed_caps = gst_caps_ref (caps); } GST_INFO_OBJECT (osssrc, "returning caps %" GST_PTR_FORMAT, caps); if (filter && caps) { GstCaps *intersection; intersection = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); return intersection; } else { return caps; } } static gint ilog2 (gint x) { /* well... hacker's delight explains... */ x = x | (x >> 1); x = x | (x >> 2); x = x | (x >> 4); x = x | (x >> 8); x = x | (x >> 16); x = x - ((x >> 1) & 0x55555555); x = (x & 0x33333333) + ((x >> 2) & 0x33333333); x = (x + (x >> 4)) & 0x0f0f0f0f; x = x + (x >> 8); x = x + (x >> 16); return (x & 0x0000003f) - 1; } static gint gst_oss_src_get_format (GstAudioRingBufferFormatType fmt, GstAudioFormat rfmt) { gint result; switch (fmt) { case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW: result = AFMT_MU_LAW; break; case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW: result = AFMT_A_LAW; break; case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM: result = AFMT_IMA_ADPCM; break; case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG: result = AFMT_MPEG; break; case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW: { switch (rfmt) { case GST_AUDIO_FORMAT_U8: result = AFMT_U8; break; case GST_AUDIO_FORMAT_S16LE: result = AFMT_S16_LE; break; case GST_AUDIO_FORMAT_S16BE: result = AFMT_S16_BE; break; case GST_AUDIO_FORMAT_S8: result = AFMT_S8; break; case GST_AUDIO_FORMAT_U16LE: result = AFMT_U16_LE; break; case GST_AUDIO_FORMAT_U16BE: result = AFMT_U16_BE; break; default: result = 0; break; } break; } default: result = 0; break; } return result; } static gboolean gst_oss_src_open (GstAudioSrc * asrc) { GstOssSrc *oss; int mode; oss = GST_OSS_SRC (asrc); mode = O_RDONLY; mode |= O_NONBLOCK; oss->fd = open (oss->device, mode, 0); if (oss->fd == -1) { switch (errno) { case EACCES: goto no_permission; default: goto open_failed; } } g_free (oss->device_name); oss->device_name = gst_oss_helper_get_card_name ("/dev/mixer"); return TRUE; no_permission: { GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, (_("Could not open audio device for recording. " "You don't have permission to open the device.")), GST_ERROR_SYSTEM); return FALSE; } open_failed: { GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, (_("Could not open audio device for recording.")), ("Unable to open device %s for recording: %s", oss->device, g_strerror (errno))); return FALSE; } } static gboolean gst_oss_src_close (GstAudioSrc * asrc) { GstOssSrc *oss; oss = GST_OSS_SRC (asrc); close (oss->fd); gst_caps_replace (&oss->probed_caps, NULL); return TRUE; } static gboolean gst_oss_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec) { GstOssSrc *oss; struct audio_buf_info info; int mode; int fmt, tmp; guint width, rate, channels; oss = GST_OSS_SRC (asrc); mode = fcntl (oss->fd, F_GETFL); mode &= ~O_NONBLOCK; if (fcntl (oss->fd, F_SETFL, mode) == -1) goto non_block; fmt = gst_oss_src_get_format (spec->type, GST_AUDIO_INFO_FORMAT (&spec->info)); if (fmt == 0) goto wrong_format; width = GST_AUDIO_INFO_WIDTH (&spec->info); rate = GST_AUDIO_INFO_RATE (&spec->info); channels = GST_AUDIO_INFO_CHANNELS (&spec->info); if (width != 16 && width != 8) goto dodgy_width; tmp = ilog2 (spec->segsize); tmp = ((spec->segtotal & 0x7fff) << 16) | tmp; GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x", spec->segsize, spec->segtotal, tmp); SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT"); SET_PARAM (oss, SNDCTL_DSP_RESET, 0, "RESET"); SET_PARAM (oss, SNDCTL_DSP_SETFMT, fmt, "SETFMT"); if (channels == 2) SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO"); SET_PARAM (oss, SNDCTL_DSP_CHANNELS, channels, "CHANNELS"); SET_PARAM (oss, SNDCTL_DSP_SPEED, rate, "SPEED"); GET_PARAM (oss, SNDCTL_DSP_GETISPACE, &info, "GETISPACE"); spec->segsize = info.fragsize; spec->segtotal = info.fragstotal; oss->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info); GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x", spec->segsize, spec->segtotal, tmp); return TRUE; non_block: { GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, ("Unable to set device %s in non blocking mode: %s", oss->device, g_strerror (errno)), (NULL)); return FALSE; } wrong_format: { GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, ("Unable to get format (%d, %d)", spec->type, GST_AUDIO_INFO_FORMAT (&spec->info)), (NULL)); return FALSE; } dodgy_width: { GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_READ, ("Unexpected width %d", width), (NULL)); return FALSE; } } static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc) { /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */ if (!gst_oss_src_close (asrc)) goto couldnt_close; if (!gst_oss_src_open (asrc)) goto couldnt_reopen; return TRUE; couldnt_close: { GST_DEBUG_OBJECT (asrc, "Could not close the audio device"); return FALSE; } couldnt_reopen: { GST_DEBUG_OBJECT (asrc, "Could not reopen the audio device"); return FALSE; } } static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length) { 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 */ }