mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-25 03:01:03 +00:00
32ba17cd0f
where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there.
525 lines
14 KiB
C
525 lines
14 KiB
C
/*
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* Copyright (C) <2008> Jacob Meuser <jakemsr@sdf.lonestar.org>
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*
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* Permission to use, copy, modify, and distribute this software for any
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* purpose with or without fee is hereby granted, provided that the above
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* copyright notice and this permission notice appear in all copies.
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*
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* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
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* WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
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* ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
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* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
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* ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
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* OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
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*/
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/**
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* SECTION:element-sndiosrc
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* @see_also: #GstAutoAudioSrc
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*
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* <refsect2>
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* <para>
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* This element retrieves samples from a sound card using sndio.
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* </para>
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* <para>
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* Simple example pipeline that records an Ogg/Vorbis file via sndio:
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* <programlisting>
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* gst-launch -v sndiosrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=foo.ogg
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* </programlisting>
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* </para>
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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 "sndiosrc.h"
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#include <unistd.h>
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#include <errno.h>
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#include <gst/gst-i18n-plugin.h>
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GST_DEBUG_CATEGORY_EXTERN (gst_sndio_debug);
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#define GST_CAT_DEFAULT gst_sndio_debug
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enum
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{
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PROP_0,
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PROP_HOST
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};
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static GstStaticPadTemplate sndio_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) { 1234, 4321 }, "
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"signed = (boolean) { TRUE, FALSE }, "
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"width = (int) { 8, 16, 24, 32 }, "
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"depth = (int) { 8, 16, 24, 32 }, "
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"rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 16 ] ")
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);
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static void gst_sndiosrc_finalize (GObject * object);
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static GstCaps *gst_sndiosrc_getcaps (GstBaseSrc * bsrc);
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static gboolean gst_sndiosrc_open (GstAudioSrc * asrc);
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static gboolean gst_sndiosrc_close (GstAudioSrc * asrc);
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static gboolean gst_sndiosrc_prepare (GstAudioSrc * asrc,
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GstRingBufferSpec * spec);
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static gboolean gst_sndiosrc_unprepare (GstAudioSrc * asrc);
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static guint gst_sndiosrc_read (GstAudioSrc * asrc, gpointer data,
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guint length);
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static guint gst_sndiosrc_delay (GstAudioSrc * asrc);
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static void gst_sndiosrc_reset (GstAudioSrc * asrc);
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static void gst_sndiosrc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_sndiosrc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_sndiosrc_cb (void *addr, int delta);
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GST_BOILERPLATE (GstSndioSrc, gst_sndiosrc, GstAudioSrc, GST_TYPE_AUDIO_SRC);
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static void
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gst_sndiosrc_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_static_metadata (element_class,
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"Sndio audio source",
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"Source/Audio",
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"Records audio through sndio", "Jacob Meuser <jakemsr@sdf.lonestar.org>");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sndio_src_factory));
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}
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static void
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gst_sndiosrc_class_init (GstSndioSrcClass * klass)
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{
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GObjectClass *gobject_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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gstbasesrc_class = (GstBaseSrcClass *) klass;
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gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
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gstaudiosrc_class = (GstAudioSrcClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_sndiosrc_finalize;
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_sndiosrc_getcaps);
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gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_sndiosrc_open);
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gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_sndiosrc_close);
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gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_sndiosrc_prepare);
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gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_sndiosrc_unprepare);
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gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_sndiosrc_read);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_sndiosrc_delay);
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gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_sndiosrc_reset);
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gobject_class->set_property = gst_sndiosrc_set_property;
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gobject_class->get_property = gst_sndiosrc_get_property;
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/* default value is filled in the _init method */
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g_object_class_install_property (gobject_class, PROP_HOST,
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g_param_spec_string ("host", "Host",
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"Device or socket sndio will access", NULL, G_PARAM_READWRITE));
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}
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static void
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gst_sndiosrc_init (GstSndioSrc * sndiosrc, GstSndioSrcClass * klass)
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{
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sndiosrc->hdl = NULL;
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sndiosrc->host = g_strdup (g_getenv ("AUDIODEVICE"));
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}
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static void
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gst_sndiosrc_finalize (GObject * object)
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{
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GstSndioSrc *sndiosrc = GST_SNDIOSRC (object);
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gst_caps_replace (&sndiosrc->cur_caps, NULL);
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g_free (sndiosrc->host);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static GstCaps *
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gst_sndiosrc_getcaps (GstBaseSrc * bsrc)
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{
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GstSndioSrc *sndiosrc;
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sndiosrc = GST_SNDIOSRC (bsrc);
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/* no hdl, we're done with the template caps */
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if (sndiosrc->cur_caps == NULL) {
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GST_LOG_OBJECT (sndiosrc, "getcaps called, returning template caps");
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return NULL;
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}
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GST_LOG_OBJECT (sndiosrc, "returning %" GST_PTR_FORMAT, sndiosrc->cur_caps);
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return gst_caps_ref (sndiosrc->cur_caps);
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}
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static gboolean
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gst_sndiosrc_open (GstAudioSrc * asrc)
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{
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GstPadTemplate *pad_template;
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GstSndioSrc *sndiosrc;
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struct sio_par par;
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struct sio_cap cap;
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GArray *rates, *chans;
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GValue rates_v = { 0 };
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GValue chans_v = { 0 };
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GValue value = { 0 };
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struct sio_enc enc;
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struct sio_conf conf;
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int confs[SIO_NCONF];
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int rate, chan;
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int i, j, k;
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int nconfs;
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sndiosrc = GST_SNDIOSRC (asrc);
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GST_DEBUG_OBJECT (sndiosrc, "open");
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/* connect */
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sndiosrc->hdl = sio_open (sndiosrc->host, SIO_REC, 0);
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if (sndiosrc->hdl == NULL)
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goto couldnt_connect;
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/* Use sndio defaults as the only encodings, but get the supported
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* sample rates and number of channels.
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*/
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if (!sio_getpar (sndiosrc->hdl, &par))
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goto no_server_info;
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if (!sio_getcap (sndiosrc->hdl, &cap))
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goto no_server_info;
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rates = g_array_new (FALSE, FALSE, sizeof (int));
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chans = g_array_new (FALSE, FALSE, sizeof (int));
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/* find confs that have the default encoding */
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nconfs = 0;
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for (i = 0; i < cap.nconf; i++) {
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for (j = 0; j < SIO_NENC; j++) {
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if (cap.confs[i].enc & (1 << j)) {
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enc = cap.enc[j];
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if (enc.bits == par.bits && enc.sig == par.sig && enc.le == par.le) {
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confs[nconfs] = i;
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nconfs++;
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break;
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}
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}
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}
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}
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/* find the rates and channels of the confs that have the default encoding */
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for (i = 0; i < nconfs; i++) {
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conf = cap.confs[confs[i]];
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/* rates */
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for (j = 0; j < SIO_NRATE; j++) {
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if (conf.rate & (1 << j)) {
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rate = cap.rate[j];
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for (k = 0; k < rates->len && rate; k++) {
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if (rate == g_array_index (rates, int, k))
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rate = 0;
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}
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/* add in ascending order */
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if (rate) {
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for (k = 0; k < rates->len; k++) {
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if (rate < g_array_index (rates, int, k))
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{
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g_array_insert_val (rates, k, rate);
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break;
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}
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}
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if (k == rates->len)
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g_array_append_val (rates, rate);
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}
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}
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}
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/* channels */
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for (j = 0; j < SIO_NCHAN; j++) {
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if (conf.rchan & (1 << j)) {
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chan = cap.rchan[j];
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for (k = 0; k < chans->len && chan; k++) {
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if (chan == g_array_index (chans, int, k))
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chan = 0;
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}
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/* add in ascending order */
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if (chan) {
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for (k = 0; k < chans->len; k++) {
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if (chan < g_array_index (chans, int, k))
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{
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g_array_insert_val (chans, k, chan);
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break;
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}
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}
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if (k == chans->len)
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g_array_append_val (chans, chan);
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}
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}
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}
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}
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/* not sure how this can happen, but it might */
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if (cap.nconf == 0) {
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g_array_append_val (rates, par.rate);
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g_array_append_val (chans, par.rchan);
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}
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g_value_init (&rates_v, GST_TYPE_LIST);
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g_value_init (&chans_v, GST_TYPE_LIST);
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g_value_init (&value, G_TYPE_INT);
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for (i = 0; i < rates->len; i++) {
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g_value_set_int (&value, g_array_index (rates, int, i));
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gst_value_list_append_value (&rates_v, &value);
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}
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for (i = 0; i < chans->len; i++) {
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g_value_set_int (&value, g_array_index (chans, int, i));
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gst_value_list_append_value (&chans_v, &value);
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}
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g_array_free (rates, TRUE);
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g_array_free (chans, TRUE);
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pad_template = gst_static_pad_template_get (&sndio_src_factory);
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sndiosrc->cur_caps = gst_caps_copy (gst_pad_template_get_caps (pad_template));
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gst_object_unref (pad_template);
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for (i = 0; i < sndiosrc->cur_caps->structs->len; i++) {
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GstStructure *s;
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s = gst_caps_get_structure (sndiosrc->cur_caps, i);
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gst_structure_set (s, "endianness", G_TYPE_INT, par.le ? 1234 : 4321, NULL);
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gst_structure_set (s, "signed", G_TYPE_BOOLEAN, par.sig ? TRUE : FALSE,
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NULL);
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gst_structure_set (s, "width", G_TYPE_INT, par.bits, NULL);
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// gst_structure_set (s, "depth", G_TYPE_INT, par.bps * 8, NULL); /* XXX */
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gst_structure_set_value (s, "rate", &rates_v);
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gst_structure_set_value (s, "channels", &chans_v);
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}
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return TRUE;
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/* ERRORS */
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couldnt_connect:
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{
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GST_ELEMENT_ERROR (sndiosrc, RESOURCE, OPEN_READ,
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(_("Could not establish connection to sndio")),
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("can't open connection to sndio"));
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return FALSE;
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}
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no_server_info:
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{
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GST_ELEMENT_ERROR (sndiosrc, RESOURCE, OPEN_READ,
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(_("Failed to query sndio capabilities")),
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("couldn't get sndio info!"));
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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_sndiosrc_close (GstAudioSrc * asrc)
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{
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GstSndioSrc *sndiosrc = GST_SNDIOSRC (asrc);
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GST_DEBUG_OBJECT (sndiosrc, "close");
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gst_caps_replace (&sndiosrc->cur_caps, NULL);
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sio_close (sndiosrc->hdl);
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sndiosrc->hdl = NULL;
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return TRUE;
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}
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static void
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gst_sndiosrc_cb (void *addr, int delta)
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{
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GstSndioSrc *sndiosrc = GST_SNDIOSRC ((GstAudioSrc *) addr);
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sndiosrc->realpos += delta;
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if (sndiosrc->readpos >= sndiosrc->realpos)
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sndiosrc->latency = 0;
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else
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sndiosrc->latency = sndiosrc->realpos - sndiosrc->readpos;
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}
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static gboolean
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gst_sndiosrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
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{
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GstSndioSrc *sndiosrc = GST_SNDIOSRC (asrc);
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struct sio_par par;
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int spec_bpf;
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GST_DEBUG_OBJECT (sndiosrc, "prepare");
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sndiosrc->readpos = sndiosrc->realpos = sndiosrc->latency = 0;
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sio_initpar (&par);
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par.sig = spec->sign;
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par.le = !spec->bigend;
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par.bits = spec->width;
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// par.bps = spec->depth / 8; /* XXX */
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par.rate = spec->rate;
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par.rchan = spec->channels;
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spec_bpf = ((spec->width / 8) * spec->channels);
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par.round = spec->segsize / spec_bpf;
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par.appbufsz = (spec->segsize * spec->segtotal) / spec_bpf;
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if (!sio_setpar (sndiosrc->hdl, &par))
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goto cannot_configure;
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sio_getpar (sndiosrc->hdl, &par);
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spec->sign = par.sig;
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spec->bigend = !par.le;
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spec->width = par.bits;
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// spec->depth = par.bps * 8; /* XXX */
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spec->rate = par.rate;
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spec->channels = par.rchan;
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sndiosrc->bpf = par.bps * par.rchan;
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spec->segsize = par.round * par.rchan * par.bps;
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spec->segtotal = par.bufsz / par.round;
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/* FIXME: this is wrong for signed ints (and the
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* audioringbuffers should do it for us anyway) */
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spec->silence_sample[0] = 0;
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spec->silence_sample[1] = 0;
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spec->silence_sample[2] = 0;
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spec->silence_sample[3] = 0;
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sio_onmove (sndiosrc->hdl, gst_sndiosrc_cb, sndiosrc);
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if (!sio_start (sndiosrc->hdl))
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goto cannot_start;
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GST_INFO_OBJECT (sndiosrc, "successfully opened connection to sndio");
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return TRUE;
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/* ERRORS */
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cannot_configure:
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{
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GST_ELEMENT_ERROR (sndiosrc, RESOURCE, OPEN_READ,
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(_("Could not configure sndio")), ("can't configure sndio"));
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return FALSE;
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}
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cannot_start:
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{
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GST_ELEMENT_ERROR (sndiosrc, RESOURCE, OPEN_READ,
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(_("Could not start sndio")), ("can't start sndio"));
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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_sndiosrc_unprepare (GstAudioSrc * asrc)
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{
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GstSndioSrc *sndiosrc = GST_SNDIOSRC (asrc);
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if (sndiosrc->hdl == NULL)
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return TRUE;
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sio_stop (sndiosrc->hdl);
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return TRUE;
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}
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static guint
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gst_sndiosrc_read (GstAudioSrc * asrc, gpointer data, guint length)
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{
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GstSndioSrc *sndiosrc = GST_SNDIOSRC (asrc);
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guint done;
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done = sio_read (sndiosrc->hdl, data, length);
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if (done == 0)
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goto read_error;
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sndiosrc->readpos += (done / sndiosrc->bpf);
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data = (char *) data + done;
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return done;
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/* ERRORS */
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read_error:
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{
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GST_ELEMENT_ERROR (sndiosrc, RESOURCE, READ,
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("Failed to read data from sndio"), GST_ERROR_SYSTEM);
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return 0;
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}
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}
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static guint
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gst_sndiosrc_delay (GstAudioSrc * asrc)
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{
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GstSndioSrc *sndiosrc = GST_SNDIOSRC (asrc);
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if (sndiosrc->latency == (guint) - 1) {
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GST_WARNING_OBJECT (asrc, "couldn't get latency");
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return 0;
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}
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GST_DEBUG_OBJECT (asrc, "got latency: %u", sndiosrc->latency);
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return sndiosrc->latency;
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}
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|
|
static void
|
|
gst_sndiosrc_reset (GstAudioSrc * asrc)
|
|
{
|
|
/* no way to flush the buffers with sndio ? */
|
|
|
|
GST_DEBUG_OBJECT (asrc, "reset called");
|
|
}
|
|
|
|
static void
|
|
gst_sndiosrc_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstSndioSrc *sndiosrc = GST_SNDIOSRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_HOST:
|
|
g_free (sndiosrc->host);
|
|
sndiosrc->host = g_value_dup_string (value);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_sndiosrc_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstSndioSrc *sndiosrc = GST_SNDIOSRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_HOST:
|
|
g_value_set_string (value, sndiosrc->host);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|