mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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36c2fc253b
Original commit message from CVS: * examples/indexing/indexmpeg.c: (main): * ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio), (gst_artsdsink_close_audio), (gst_artsdsink_change_state): * ext/artsd/gstartsdsink.h: * ext/audiofile/gstafparse.c: (gst_afparse_open_file), (gst_afparse_close_file): * ext/audiofile/gstafparse.h: * ext/audiofile/gstafsink.c: (gst_afsink_open_file), (gst_afsink_close_file), (gst_afsink_chain), (gst_afsink_change_state): * ext/audiofile/gstafsink.h: * ext/audiofile/gstafsrc.c: (gst_afsrc_open_file), (gst_afsrc_close_file), (gst_afsrc_change_state): * ext/audiofile/gstafsrc.h: * ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init): * ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init): * ext/dts/gstdtsdec.c: (gst_dtsdec_init): * ext/jack/gstjack.h: * ext/jack/gstjackbin.c: (gst_jack_bin_init), (gst_jack_bin_change_state): * ext/musepack/gstmusepackdec.c: (gst_musepackdec_init): * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init): * ext/nas/nassink.c: (gst_nassink_open_audio), (gst_nassink_close_audio), (gst_nassink_change_state): * ext/nas/nassink.h: * ext/polyp/polypsink.c: (gst_polypsink_init): * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state): * ext/sdl/sdlvideosink.h: * ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init): * ext/sndfile/gstsf.c: (gst_sf_set_property), (gst_sf_change_state), (gst_sf_release_request_pad), (gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop): * ext/sndfile/gstsf.h: * ext/swfdec/gstswfdec.c: (gst_swfdec_init): * ext/tarkin/gsttarkindec.c: (gst_tarkindec_init): * gst/apetag/apedemux.c: (gst_ape_demux_init): * gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init): * gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init): * gst/festival/gstfestival.c: (gst_festival_change_state): * gst/festival/gstfestival.h: * gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init): * gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init), (gst_multifilesink_set_location), (gst_multifilesink_open_file), (gst_multifilesink_close_file), (gst_multifilesink_next_file), (gst_multifilesink_pad_query), (gst_multifilesink_handle_event), (gst_multifilesink_chain), (gst_multifilesink_change_state): * gst/multifilesink/gstmultifilesink.h: * gst/videodrop/gstvideodrop.c: (gst_videodrop_init): * sys/cdrom/gstcdplayer.c: (cdplayer_init): * sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init), (dxr3audiosink_open), (dxr3audiosink_close), (dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3), (dxr3audiosink_change_state): * sys/dxr3/dxr3audiosink.h: * sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open), (dxr3spusink_close), (dxr3spusink_chain), (dxr3spusink_change_state): * sys/dxr3/dxr3spusink.h: * sys/dxr3/dxr3videosink.c: (dxr3videosink_init), (dxr3videosink_open), (dxr3videosink_close), (dxr3videosink_write_data), (dxr3videosink_change_state): * sys/dxr3/dxr3videosink.h: * sys/glsink/glimagesink.c: (gst_glimagesink_init): * sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state), (gst_qcamsrc_open), (gst_qcamsrc_close): * sys/qcam/gstqcamsrc.h: * sys/v4l2/gstv4l2src.c: (gst_v4l2src_init): * sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get), (gst_vcdsrc_open_file), (gst_vcdsrc_close_file), (gst_vcdsrc_change_state), (gst_vcdsrc_recalculate): * sys/vcd/vcdsrc.h: renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition
880 lines
24 KiB
C
880 lines
24 KiB
C
/* GStreamer libsndfile plugin
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* Copyright (C) 2003 Andy Wingo <wingo at pobox dot com>
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gst/gst-i18n-plugin.h"
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#include <string.h>
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include "gstsf.h"
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static GstElementDetails sfsrc_details = {
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"Sndfile Source",
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"Source/Audio",
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"Read audio streams from disk using libsndfile",
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"Andy Wingo <wingo at pobox dot com>",
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};
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static GstElementDetails sfsink_details = {
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"Sndfile Sink",
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"Sink/Audio",
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"Write audio streams to disk using libsndfile",
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"Andy Wingo <wingo at pobox dot com>",
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};
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enum
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{
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ARG_0,
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ARG_LOCATION,
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ARG_MAJOR_TYPE,
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ARG_MINOR_TYPE,
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ARG_LOOP,
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ARG_CREATE_PADS
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};
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static GstStaticPadTemplate sf_src_factory = GST_STATIC_PAD_TEMPLATE ("src%d",
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GST_PAD_SRC,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS)
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);
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static GstStaticPadTemplate sf_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink%d",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS)
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);
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#define GST_TYPE_SF_MAJOR_TYPES (gst_sf_major_types_get_type())
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static GType
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gst_sf_major_types_get_type (void)
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{
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static GType sf_major_types_type = 0;
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static GEnumValue *sf_major_types = NULL;
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if (!sf_major_types_type) {
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SF_FORMAT_INFO format_info;
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int k, count;
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sf_command (NULL, SFC_GET_FORMAT_MAJOR_COUNT, &count, sizeof (int));
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sf_major_types = g_new0 (GEnumValue, count + 1);
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for (k = 0; k < count; k++) {
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format_info.format = k;
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sf_command (NULL, SFC_GET_FORMAT_MAJOR, &format_info,
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sizeof (format_info));
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sf_major_types[k].value = format_info.format;
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sf_major_types[k].value_name = g_strdup (format_info.name);
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sf_major_types[k].value_nick = g_strdup (format_info.extension);
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/* Irritatingly enough, there exist major_types with the same extension. Let's
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just hope that sndfile gives us the list in alphabetical order, as it
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currently does. */
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if (k > 0
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&& strcmp (sf_major_types[k].value_nick,
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sf_major_types[k - 1].value_nick) == 0) {
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g_free (sf_major_types[k].value_nick);
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sf_major_types[k].value_nick =
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g_strconcat (sf_major_types[k - 1].value_nick, "-",
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sf_major_types[k].value_name, NULL);
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g_strcanon (sf_major_types[k].value_nick,
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G_CSET_A_2_Z G_CSET_a_2_z G_CSET_DIGITS "-", '-');
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}
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}
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sf_major_types_type =
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g_enum_register_static ("GstSndfileMajorTypes", sf_major_types);
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}
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return sf_major_types_type;
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}
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#define GST_TYPE_SF_MINOR_TYPES (gst_sf_minor_types_get_type())
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static GType
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gst_sf_minor_types_get_type (void)
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{
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static GType sf_minor_types_type = 0;
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static GEnumValue *sf_minor_types = NULL;
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if (!sf_minor_types_type) {
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SF_FORMAT_INFO format_info;
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int k, count;
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sf_command (NULL, SFC_GET_FORMAT_SUBTYPE_COUNT, &count, sizeof (int));
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sf_minor_types = g_new0 (GEnumValue, count + 1);
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for (k = 0; k < count; k++) {
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format_info.format = k;
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sf_command (NULL, SFC_GET_FORMAT_SUBTYPE, &format_info,
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sizeof (format_info));
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sf_minor_types[k].value = format_info.format;
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sf_minor_types[k].value_name = g_strdup (format_info.name);
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sf_minor_types[k].value_nick = g_ascii_strdown (format_info.name, -1);
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g_strcanon (sf_minor_types[k].value_nick, G_CSET_a_2_z G_CSET_DIGITS "-",
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'-');
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}
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sf_minor_types_type =
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g_enum_register_static ("GstSndfileMinorTypes", sf_minor_types);
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}
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return sf_minor_types_type;
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}
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static void gst_sfsrc_base_init (gpointer g_class);
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static void gst_sfsink_base_init (gpointer g_class);
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static void gst_sf_class_init (GstSFClass * klass);
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static void gst_sf_init (GstSF * this);
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static void gst_sf_dispose (GObject * object);
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static void gst_sf_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_sf_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstClock *gst_sf_get_clock (GstElement * element);
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static void gst_sf_set_clock (GstElement * element, GstClock * clock);
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static GstPad *gst_sf_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * unused);
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static void gst_sf_release_request_pad (GstElement * element, GstPad * pad);
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static GstStateChangeReturn gst_sf_change_state (GstElement * element,
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GstStateChange transition);
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static GstPadLinkReturn gst_sf_link (GstPad * pad, const GstCaps * caps);
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static void gst_sf_loop (GstElement * element);
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static GstClockTime gst_sf_get_time (GstClock * clock, gpointer data);
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static gboolean gst_sf_open_file (GstSF * this);
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static void gst_sf_close_file (GstSF * this);
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static GstElementClass *parent_class = NULL;
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GST_DEBUG_CATEGORY_STATIC (gstsf_debug);
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#define INFO(...) \
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GST_CAT_LEVEL_LOG (gstsf_debug, GST_LEVEL_INFO, NULL, __VA_ARGS__)
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#define INFO_OBJ(obj,...) \
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GST_CAT_LEVEL_LOG (gstsf_debug, GST_LEVEL_INFO, obj, __VA_ARGS__)
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GType
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gst_sf_get_type (void)
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{
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static GType sf_type = 0;
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if (!sf_type) {
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static const GTypeInfo sf_info = {
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sizeof (GstSFClass), NULL,
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NULL,
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(GClassInitFunc) NULL, /* don't even initialize the class */
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NULL,
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NULL,
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sizeof (GstSF),
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0,
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(GInstanceInitFunc) NULL /* abstract base class */
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};
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sf_type = g_type_register_static (GST_TYPE_ELEMENT, "GstSF", &sf_info, 0);
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}
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return sf_type;
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}
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GType
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gst_sfsrc_get_type (void)
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{
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static GType sfsrc_type = 0;
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if (!sfsrc_type) {
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static const GTypeInfo sfsrc_info = {
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sizeof (GstSFClass),
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gst_sfsrc_base_init,
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NULL,
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(GClassInitFunc) gst_sf_class_init,
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NULL,
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NULL,
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sizeof (GstSF),
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0,
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(GInstanceInitFunc) gst_sf_init,
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};
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sfsrc_type =
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g_type_register_static (GST_TYPE_SF, "GstSFSrc", &sfsrc_info, 0);
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}
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return sfsrc_type;
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}
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GType
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gst_sfsink_get_type (void)
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{
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static GType sfsink_type = 0;
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if (!sfsink_type) {
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static const GTypeInfo sfsink_info = {
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sizeof (GstSFClass),
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gst_sfsink_base_init,
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NULL,
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(GClassInitFunc) gst_sf_class_init,
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NULL,
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NULL,
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sizeof (GstSF),
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0,
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(GInstanceInitFunc) gst_sf_init,
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};
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sfsink_type =
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g_type_register_static (GST_TYPE_SF, "GstSFSink", &sfsink_info, 0);
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}
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return sfsink_type;
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}
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static void
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gst_sfsrc_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_add_pad_template (element_class,
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gst_static_pad_template_get (&sf_src_factory));
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gst_element_class_set_details (element_class, &sfsrc_details);
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}
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static void
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gst_sfsink_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_add_pad_template (element_class,
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gst_static_pad_template_get (&sf_sink_factory));
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gst_element_class_set_details (element_class, &sfsink_details);
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}
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static void
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gst_sf_class_init (GstSFClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GParamSpec *pspec;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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/* although this isn't really the parent class, that's ok; GstSF doesn't
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override any methods */
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gst_element_class_install_std_props (gstelement_class, "location",
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ARG_LOCATION, G_PARAM_READWRITE, NULL);
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pspec = g_param_spec_enum
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("major-type", "Major type", "Major output type", GST_TYPE_SF_MAJOR_TYPES,
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SF_FORMAT_WAV, G_PARAM_READWRITE | G_PARAM_CONSTRUCT);
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g_object_class_install_property (gobject_class, ARG_MAJOR_TYPE, pspec);
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pspec = g_param_spec_enum
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("minor-type", "Minor type", "Minor output type", GST_TYPE_SF_MINOR_TYPES,
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SF_FORMAT_FLOAT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT);
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g_object_class_install_property (gobject_class, ARG_MINOR_TYPE, pspec);
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if (G_TYPE_FROM_CLASS (klass) == GST_TYPE_SFSRC) {
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pspec = g_param_spec_boolean ("loop", "Loop?", "Loop the output?",
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FALSE, G_PARAM_READWRITE | G_PARAM_CONSTRUCT);
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g_object_class_install_property (gobject_class, ARG_LOOP, pspec);
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pspec =
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g_param_spec_boolean ("create-pads", "Create pads?",
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"Create one pad for each channel in the sound file?", TRUE,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT);
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g_object_class_install_property (gobject_class, ARG_CREATE_PADS, pspec);
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}
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gobject_class->dispose = gst_sf_dispose;
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gobject_class->set_property = gst_sf_set_property;
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gobject_class->get_property = gst_sf_get_property;
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gstelement_class->get_clock = gst_sf_get_clock;
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gstelement_class->set_clock = gst_sf_set_clock;
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gstelement_class->change_state = gst_sf_change_state;
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gstelement_class->request_new_pad = gst_sf_request_new_pad;
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gstelement_class->release_pad = gst_sf_release_request_pad;
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}
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static void
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gst_sf_init (GstSF * this)
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{
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gst_element_set_loop_function (GST_ELEMENT (this), gst_sf_loop);
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this->provided_clock = gst_audio_clock_new ("sfclock", gst_sf_get_time, this);
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gst_object_set_parent (GST_OBJECT (this->provided_clock), GST_OBJECT (this));
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}
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static void
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gst_sf_dispose (GObject * object)
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{
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GstSF *this = (GstSF *) object;
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if (this->provided_clock) {
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gst_object_unparent (GST_OBJECT (this->provided_clock));
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this->provided_clock = 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_sf_set_property (GObject * object, guint prop_id, const GValue * value,
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GParamSpec * pspec)
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{
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GstSF *this = GST_SF (object);
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switch (prop_id) {
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case ARG_LOCATION:
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if (GST_OBJECT_FLAG_IS_SET (object, GST_SF_OPEN))
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gst_sf_close_file (this);
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if (this->filename)
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g_free (this->filename);
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if (g_value_get_string (value))
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this->filename = g_strdup (g_value_get_string (value));
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else
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this->filename = NULL;
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if (this->filename)
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gst_sf_open_file (this);
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break;
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case ARG_MAJOR_TYPE:
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this->format_major = g_value_get_enum (value);
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break;
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case ARG_MINOR_TYPE:
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this->format_subtype = g_value_get_enum (value);
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break;
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case ARG_LOOP:
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this->loop = g_value_get_boolean (value);
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break;
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case ARG_CREATE_PADS:
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this->create_pads = g_value_get_boolean (value);
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if (this->file && this->create_pads) {
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int i;
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for (i = g_list_length (this->channels); i < this->numchannels; i++)
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gst_element_get_request_pad ((GstElement *) this, "src%d");
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}
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|
break;
|
|
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_sf_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstSF *this = GST_SF (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_LOCATION:
|
|
g_value_set_string (value, this->filename);
|
|
break;
|
|
|
|
case ARG_MAJOR_TYPE:
|
|
g_value_set_enum (value, this->format_major);
|
|
break;
|
|
|
|
case ARG_MINOR_TYPE:
|
|
g_value_set_enum (value, this->format_subtype);
|
|
break;
|
|
|
|
case ARG_LOOP:
|
|
g_value_set_boolean (value, this->loop);
|
|
break;
|
|
|
|
case ARG_CREATE_PADS:
|
|
g_value_set_boolean (value, this->create_pads);
|
|
break;
|
|
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstClock *
|
|
gst_sf_get_clock (GstElement * element)
|
|
{
|
|
GstSF *this = GST_SF (element);
|
|
|
|
return this->provided_clock;
|
|
}
|
|
|
|
static void
|
|
gst_sf_set_clock (GstElement * element, GstClock * clock)
|
|
{
|
|
GstSF *this = GST_SF (element);
|
|
|
|
this->clock = clock;
|
|
}
|
|
|
|
static GstClockTime
|
|
gst_sf_get_time (GstClock * clock, gpointer data)
|
|
{
|
|
GstSF *this = GST_SF (data);
|
|
|
|
return this->time;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_sf_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstSF *this = GST_SF (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
gst_audio_clock_set_active (GST_AUDIO_CLOCK (this->provided_clock), TRUE);
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
gst_audio_clock_set_active (GST_AUDIO_CLOCK (this->provided_clock),
|
|
FALSE);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
if (GST_OBJECT_FLAG_IS_SET (this, GST_SF_OPEN))
|
|
gst_sf_close_file (this);
|
|
break;
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
return GST_STATE_CHANGE_SUCCESS;
|
|
}
|
|
|
|
static GstPad *
|
|
gst_sf_request_new_pad (GstElement * element, GstPadTemplate * templ,
|
|
const gchar * unused)
|
|
{
|
|
gchar *name;
|
|
GstSF *this;
|
|
GstSFChannel *channel;
|
|
|
|
this = GST_SF (element);
|
|
channel = g_new0 (GstSFChannel, 1);
|
|
|
|
if (templ->direction == GST_PAD_SINK) {
|
|
/* we have an SFSink */
|
|
name = g_strdup_printf ("sink%d", this->channelcount);
|
|
this->numchannels++;
|
|
if (this->file) {
|
|
gst_sf_close_file (this);
|
|
gst_sf_open_file (this);
|
|
}
|
|
} else {
|
|
/* we have an SFSrc */
|
|
name = g_strdup_printf ("src%d", this->channelcount);
|
|
}
|
|
|
|
channel->pad = gst_pad_new_from_template (templ, name);
|
|
gst_element_add_pad (GST_ELEMENT (this), channel->pad);
|
|
gst_pad_set_link_function (channel->pad, gst_sf_link);
|
|
|
|
this->channels = g_list_append (this->channels, channel);
|
|
this->channelcount++;
|
|
|
|
INFO_OBJ (element, "added pad %s\n", name);
|
|
|
|
g_free (name);
|
|
return channel->pad;
|
|
}
|
|
|
|
static void
|
|
gst_sf_release_request_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
GstSF *this;
|
|
GstSFChannel *channel = NULL;
|
|
GList *l;
|
|
|
|
this = GST_SF (element);
|
|
|
|
if (GST_STATE (element) == GST_STATE_PLAYING) {
|
|
g_warning
|
|
("You can't release a request pad if the element is PLAYING, sorry.");
|
|
return;
|
|
}
|
|
|
|
for (l = this->channels; l; l = l->next) {
|
|
if (GST_SF_CHANNEL (l)->pad == pad) {
|
|
channel = GST_SF_CHANNEL (l);
|
|
break;
|
|
}
|
|
}
|
|
|
|
g_return_if_fail (channel != NULL);
|
|
|
|
INFO_OBJ (element, "Releasing request pad %s", GST_PAD_NAME (channel->pad));
|
|
|
|
if (GST_OBJECT_FLAG_IS_SET (element, GST_SF_OPEN))
|
|
gst_sf_close_file (this);
|
|
|
|
gst_element_remove_pad (element, channel->pad);
|
|
this->channels = g_list_remove (this->channels, channel);
|
|
this->numchannels--;
|
|
g_free (channel);
|
|
}
|
|
|
|
static GstPadLinkReturn
|
|
gst_sf_link (GstPad * pad, const GstCaps * caps)
|
|
{
|
|
GstSF *this = (GstSF *) GST_OBJECT_PARENT (pad);
|
|
GstStructure *structure;
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
gst_structure_get_int (structure, "rate", &this->rate);
|
|
gst_structure_get_int (structure, "buffer-frames", &this->buffer_frames);
|
|
|
|
INFO_OBJ (this, "linked pad %s:%s with fixed caps, rate=%d, frames=%d",
|
|
GST_DEBUG_PAD_NAME (pad), this->rate, this->buffer_frames);
|
|
|
|
if (this->numchannels) {
|
|
/* we can go ahead and allocate our buffer */
|
|
if (this->buffer)
|
|
g_free (this->buffer);
|
|
this->buffer =
|
|
g_malloc (this->numchannels * this->buffer_frames * sizeof (float));
|
|
memset (this->buffer, 0,
|
|
this->numchannels * this->buffer_frames * sizeof (float));
|
|
}
|
|
return GST_PAD_LINK_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_sf_open_file (GstSF * this)
|
|
{
|
|
int mode;
|
|
SF_INFO info;
|
|
|
|
g_return_val_if_fail (!GST_OBJECT_FLAG_IS_SET (this, GST_SF_OPEN), FALSE);
|
|
|
|
this->time = 0;
|
|
|
|
if (!this->filename) {
|
|
GST_ELEMENT_ERROR (this, RESOURCE, NOT_FOUND,
|
|
(_("No filename specified.")), (NULL));
|
|
return FALSE;
|
|
}
|
|
|
|
if (GST_IS_SFSRC (this)) {
|
|
mode = SFM_READ;
|
|
info.format = 0;
|
|
} else {
|
|
if (!this->rate) {
|
|
INFO_OBJ (this, "Not opening %s yet because caps are not set",
|
|
this->filename);
|
|
return FALSE;
|
|
} else if (!this->numchannels) {
|
|
INFO_OBJ (this, "Not opening %s yet because we have no input channels",
|
|
this->filename);
|
|
return FALSE;
|
|
}
|
|
|
|
mode = SFM_WRITE;
|
|
this->format = this->format_major | this->format_subtype;
|
|
info.samplerate = this->rate;
|
|
info.channels = this->numchannels;
|
|
info.format = this->format;
|
|
|
|
INFO_OBJ (this, "Opening %s with rate %d, %d channels, format 0x%x",
|
|
this->filename, info.samplerate, info.channels, info.format);
|
|
|
|
if (!sf_format_check (&info)) {
|
|
GST_ELEMENT_ERROR (this, STREAM, ENCODE, (NULL),
|
|
("Input parameters (rate:%d, channels:%d, format:0x%x) invalid",
|
|
info.samplerate, info.channels, info.format));
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
this->file = sf_open (this->filename, mode, &info);
|
|
|
|
if (!this->file) {
|
|
GST_ELEMENT_ERROR (this, RESOURCE, OPEN_WRITE,
|
|
(_("Could not open file \"%s\" for writing."), this->filename),
|
|
("soundfile error: %s", sf_strerror (NULL)));
|
|
return FALSE;
|
|
}
|
|
|
|
if (GST_IS_SFSRC (this)) {
|
|
GList *l = NULL;
|
|
|
|
/* the number of channels in the file can be different than the number of
|
|
* pads */
|
|
this->numchannels = info.channels;
|
|
this->rate = info.samplerate;
|
|
|
|
if (this->create_pads) {
|
|
int i;
|
|
|
|
for (i = g_list_length (this->channels); i < this->numchannels; i++)
|
|
gst_element_get_request_pad ((GstElement *) this, "src%d");
|
|
}
|
|
|
|
for (l = this->channels; l; l = l->next)
|
|
/* queue the need to set caps */
|
|
GST_SF_CHANNEL (l)->caps_set = FALSE;
|
|
}
|
|
|
|
GST_OBJECT_FLAG_SET (this, GST_SF_OPEN);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_sf_close_file (GstSF * this)
|
|
{
|
|
int err = 0;
|
|
|
|
g_return_if_fail (GST_OBJECT_FLAG_IS_SET (this, GST_SF_OPEN));
|
|
|
|
INFO_OBJ (this, "Closing file %s", this->filename);
|
|
|
|
if ((err = sf_close (this->file)))
|
|
GST_ELEMENT_ERROR (this, RESOURCE, CLOSE,
|
|
("Could not close file file \"%s\".", this->filename),
|
|
("soundfile error: %s", strerror (err)));
|
|
else
|
|
GST_OBJECT_FLAG_UNSET (this, GST_SF_OPEN);
|
|
|
|
this->file = NULL;
|
|
if (this->buffer)
|
|
g_free (this->buffer);
|
|
this->buffer = NULL;
|
|
}
|
|
|
|
static void
|
|
gst_sf_loop (GstElement * element)
|
|
{
|
|
GstSF *this;
|
|
GList *l = NULL;
|
|
|
|
this = (GstSF *) element;
|
|
|
|
if (this->channels == NULL) {
|
|
GST_ELEMENT_ERROR (element, CORE, PAD, (NULL),
|
|
("You must connect at least one pad to sndfile elements."));
|
|
return;
|
|
}
|
|
|
|
if (GST_IS_SFSRC (this)) {
|
|
sf_count_t read;
|
|
gint i, j;
|
|
int eos = 0;
|
|
int buffer_frames = this->buffer_frames;
|
|
int nchannels = this->numchannels;
|
|
GstSFChannel *channel = NULL;
|
|
gfloat *data;
|
|
gfloat *buf = this->buffer;
|
|
GstBuffer *out;
|
|
|
|
if (!GST_OBJECT_FLAG_IS_SET (this, GST_SF_OPEN))
|
|
if (!gst_sf_open_file (this))
|
|
return; /* we've already set gst_element_error */
|
|
|
|
if (buffer_frames == 0) {
|
|
/* we have to set the caps later */
|
|
buffer_frames = this->buffer_frames = 1024;
|
|
}
|
|
if (buf == NULL) {
|
|
buf = this->buffer =
|
|
g_malloc (this->numchannels * this->buffer_frames * sizeof (float));
|
|
memset (this->buffer, 0,
|
|
this->numchannels * this->buffer_frames * sizeof (float));
|
|
}
|
|
|
|
read = sf_readf_float (this->file, buf, buffer_frames);
|
|
if (read < buffer_frames)
|
|
eos = 1;
|
|
|
|
if (read)
|
|
for (i = 0, l = this->channels; l; l = l->next, i++) {
|
|
channel = GST_SF_CHANNEL (l);
|
|
|
|
/* don't push on disconnected pads -- useful for ::create-pads=TRUE */
|
|
if (!GST_PAD_PEER (channel->pad))
|
|
continue;
|
|
|
|
if (!channel->caps_set) {
|
|
GstCaps *caps =
|
|
gst_caps_copy (GST_PAD_CAPS (GST_SF_CHANNEL (l)->pad));
|
|
if (!caps)
|
|
caps = gst_caps_copy
|
|
(GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (GST_SF_CHANNEL
|
|
(l)->pad)));
|
|
gst_caps_set_simple (caps, "rate", G_TYPE_INT, this->rate,
|
|
"buffer-frames", G_TYPE_INT, this->buffer_frames, NULL);
|
|
if (!gst_pad_try_set_caps (GST_SF_CHANNEL (l)->pad, caps)) {
|
|
GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL),
|
|
("Opened file with sample rate %d, but could not set caps",
|
|
this->rate));
|
|
gst_sf_close_file (this);
|
|
return;
|
|
}
|
|
channel->caps_set = TRUE;
|
|
}
|
|
|
|
out = gst_buffer_new_and_alloc (read * sizeof (float));
|
|
data = (gfloat *) GST_BUFFER_DATA (out);
|
|
for (j = 0; j < read; j++)
|
|
data[j] = buf[j * nchannels + i % nchannels];
|
|
gst_pad_push (channel->pad, GST_DATA (out));
|
|
}
|
|
|
|
this->time += read * (GST_SECOND / this->rate);
|
|
gst_audio_clock_update_time ((GstAudioClock *) this->provided_clock,
|
|
this->time);
|
|
|
|
if (eos) {
|
|
if (this->loop) {
|
|
sf_seek (this->file, (sf_count_t) 0, SEEK_SET);
|
|
eos = 0;
|
|
} else {
|
|
for (l = this->channels; l; l = l->next)
|
|
gst_pad_push (GST_SF_CHANNEL (l)->pad,
|
|
GST_DATA (gst_event_new (GST_EVENT_EOS)));
|
|
gst_element_set_eos (element);
|
|
}
|
|
}
|
|
} else {
|
|
sf_count_t written, num_to_write;
|
|
gint i, j;
|
|
int buffer_frames = this->buffer_frames;
|
|
int nchannels = this->numchannels;
|
|
GstSFChannel *channel = NULL;
|
|
gfloat *data;
|
|
gfloat *buf = this->buffer;
|
|
GstBuffer *in;
|
|
|
|
/* the problem: we can't allocate a buffer for pulled data before caps is
|
|
* set, and we can't open the file without the sample rate from the
|
|
* caps... */
|
|
|
|
num_to_write = buffer_frames;
|
|
|
|
INFO_OBJ (this, "looping, buffer_frames=%d, nchannels=%d", buffer_frames,
|
|
nchannels);
|
|
|
|
for (i = 0, l = this->channels; l; l = l->next, i++) {
|
|
channel = GST_SF_CHANNEL (l);
|
|
|
|
pull_again:
|
|
in = GST_BUFFER (gst_pad_pull (channel->pad));
|
|
|
|
if (buffer_frames == 0) {
|
|
/* pulling a buffer from the pad should have caused capsnego to occur,
|
|
which then would set this->buffer_frames to a new value */
|
|
buffer_frames = this->buffer_frames;
|
|
if (buffer_frames == 0) {
|
|
GST_ELEMENT_ERROR (element, CORE, NEGOTIATION, (NULL),
|
|
("format wasn't negotiated before chain function"));
|
|
return;
|
|
}
|
|
buf = this->buffer;
|
|
num_to_write = buffer_frames;
|
|
}
|
|
|
|
if (!GST_OBJECT_FLAG_IS_SET (this, GST_SF_OPEN))
|
|
if (!gst_sf_open_file (this))
|
|
return; /* we've already set gst_element_error */
|
|
|
|
if (GST_IS_EVENT (in)) {
|
|
switch (GST_EVENT_TYPE (in)) {
|
|
case GST_EVENT_EOS:
|
|
case GST_EVENT_INTERRUPT:
|
|
num_to_write = 0;
|
|
break;
|
|
default:
|
|
goto pull_again;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (num_to_write) {
|
|
data = (gfloat *) GST_BUFFER_DATA (in);
|
|
num_to_write =
|
|
MIN (num_to_write, GST_BUFFER_SIZE (in) / sizeof (gfloat));
|
|
for (j = 0; j < num_to_write; j++)
|
|
buf[j * nchannels + i % nchannels] = data[j];
|
|
}
|
|
|
|
gst_data_unref ((GstData *) in);
|
|
}
|
|
|
|
if (num_to_write) {
|
|
written = sf_writef_float (this->file, buf, num_to_write);
|
|
if (written != num_to_write)
|
|
GST_ELEMENT_ERROR (element, RESOURCE, WRITE,
|
|
(_("Could not write to file \"%s\"."), this->filename),
|
|
("soundfile error: %s", sf_strerror (this->file)));
|
|
}
|
|
|
|
this->time += num_to_write * (GST_SECOND / this->rate);
|
|
gst_audio_clock_update_time ((GstAudioClock *) this->provided_clock,
|
|
this->time);
|
|
|
|
if (num_to_write != buffer_frames)
|
|
gst_element_set_eos (element);
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
if (!gst_library_load ("gstaudio"))
|
|
return FALSE;
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gstsf_debug, "sf",
|
|
GST_DEBUG_FG_WHITE | GST_DEBUG_BG_GREEN | GST_DEBUG_BOLD,
|
|
"libsndfile plugin");
|
|
|
|
if (!gst_element_register (plugin, "sfsrc", GST_RANK_NONE, GST_TYPE_SFSRC))
|
|
return FALSE;
|
|
|
|
if (!gst_element_register (plugin, "sfsink", GST_RANK_NONE, GST_TYPE_SFSINK))
|
|
return FALSE;
|
|
|
|
#ifdef ENABLE_NLS
|
|
setlocale (LC_ALL, "");
|
|
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
|
|
#endif /* ENABLE_NLS */
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"sndfile",
|
|
"use libsndfile to read and write audio from and to files",
|
|
plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
|