gst/cutter/Makefile
gst/deinterlace/Makefile
gst/flx/Makefile
+gst/goom/Makefile
gst/intfloat/Makefile
gst/law/Makefile
gst/level/Makefile
ext/sidplay/Makefile
ext/smoothwave/Makefile
ext/vorbis/Makefile
+ext/tarkin/Makefile
ext/xmms/Makefile
gst-libs/Makefile
gst-libs/gst/Makefile
}
static void
+gst_aasink_set_clock (GstElement *element, GstClock *clock)
+{
+ GstAASink *aasink = GST_AASINK (element);
+
+ aasink->clock = clock;
+}
+
+static void
gst_aasink_init (GstAASink *aasink)
{
aasink->sinkpad = gst_pad_new_from_template (
gst_pad_set_chain_function (aasink->sinkpad, gst_aasink_chain);
gst_pad_set_connect_function (aasink->sinkpad, gst_aasink_sinkconnect);
- aasink->clock = gst_clock_get_system();
- gst_clock_register(aasink->clock, GST_OBJECT(aasink));
-
memcpy(&aasink->ascii_surf, &aa_defparams, sizeof (struct aa_hardware_params));
aasink->ascii_parms.bright = 0;
aasink->ascii_parms.contrast = 16;
aasink->width = -1;
aasink->height = -1;
+ aasink->clock = NULL;
+ GST_ELEMENT (aasink)->setclockfunc = gst_aasink_set_clock;
+
GST_FLAG_SET(aasink, GST_ELEMENT_THREAD_SUGGESTED);
}
gst_aasink_chain (GstPad *pad, GstBuffer *buf)
{
GstAASink *aasink;
- GstClockTimeDiff jitter;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
GST_DEBUG (0,"videosink: clock wait: %llu\n", GST_BUFFER_TIMESTAMP(buf));
- jitter = gst_clock_current_diff(aasink->clock, GST_BUFFER_TIMESTAMP (buf));
-
- if (jitter > 500000 || jitter < -500000)
- {
- GST_DEBUG (0, "jitter: %lld\n", jitter);
- gst_clock_set (aasink->clock, GST_BUFFER_TIMESTAMP (buf));
- }
- else {
- gst_clock_wait(aasink->clock, GST_BUFFER_TIMESTAMP(buf), GST_OBJECT(aasink));
+ if (aasink->clock) {
+ gst_element_clock_wait (GST_ELEMENT (aasink), aasink->clock, GST_BUFFER_TIMESTAMP(buf));
}
aa_render (aasink->context, &aasink->ascii_parms,
--- /dev/null
+/* Gnome-Streamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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.
+ */
+
+
+/*#define GST_DEBUG_ENABLED */
+#include <string.h>
+
+#include "gstavidecoder.h"
+
+
+
+/* elementfactory information */
+static GstElementDetails gst_avi_decoder_details = {
+ ".avi decoder",
+ "Decoder/Video",
+ "Decodes a .avi file into audio and video",
+ VERSION,
+ "Erik Walthinsen <omega@cse.ogi.edu>\n"
+ "Wim Taymans <wim.taymans@tvd.be>",
+ "(C) 1999",
+};
+
+static GstCaps* avi_typefind (GstBuffer *buf, gpointer private);
+
+/* typefactory for 'avi' */
+static GstTypeDefinition avidefinition = {
+ "avidecoder_video/avi",
+ "video/avi",
+ ".avi",
+ avi_typefind,
+};
+
+/* AviDecoder signals and args */
+enum {
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum {
+ ARG_0,
+ ARG_BITRATE,
+ ARG_MEDIA_TIME,
+ ARG_CURRENT_TIME,
+ /* FILL ME */
+};
+
+GST_PADTEMPLATE_FACTORY (sink_templ,
+ "sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "avidecoder_sink",
+ "video/avi",
+ "RIFF", GST_PROPS_STRING ("AVI")
+ )
+)
+
+GST_PADTEMPLATE_FACTORY (src_video_templ,
+ "video_src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "wincodec_src",
+ "video/raw",
+ "format", GST_PROPS_LIST (
+ GST_PROPS_FOURCC (GST_MAKE_FOURCC ('Y','U','Y','2')),
+ GST_PROPS_FOURCC (GST_MAKE_FOURCC ('I','4','2','0')),
+ GST_PROPS_FOURCC (GST_MAKE_FOURCC ('R','G','B',' '))
+ ),
+ "width", GST_PROPS_INT_RANGE (16, 4096),
+ "height", GST_PROPS_INT_RANGE (16, 4096)
+ )
+)
+
+GST_PADTEMPLATE_FACTORY (src_audio_templ,
+ "audio_src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "src_audio",
+ "audio/raw",
+ "format", GST_PROPS_STRING ("int"),
+ "law", GST_PROPS_INT (0),
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+ "signed", GST_PROPS_LIST (
+ GST_PROPS_BOOLEAN (TRUE),
+ GST_PROPS_BOOLEAN (FALSE)
+ ),
+ "width", GST_PROPS_LIST (
+ GST_PROPS_INT (8),
+ GST_PROPS_INT (16)
+ ),
+ "depth", GST_PROPS_LIST (
+ GST_PROPS_INT (8),
+ GST_PROPS_INT (16)
+ ),
+ "rate", GST_PROPS_INT_RANGE (11025, 48000),
+ "channels", GST_PROPS_INT_RANGE (1, 2)
+ )
+)
+
+static void gst_avi_decoder_class_init (GstAviDecoderClass *klass);
+static void gst_avi_decoder_init (GstAviDecoder *avi_decoder);
+
+static void gst_avi_decoder_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
+
+
+
+static GstElementClass *parent_class = NULL;
+/*static guint gst_avi_decoder_signals[LAST_SIGNAL] = { 0 }; */
+
+GType
+gst_avi_decoder_get_type(void)
+{
+ static GType avi_decoder_type = 0;
+
+ if (!avi_decoder_type) {
+ static const GTypeInfo avi_decoder_info = {
+ sizeof(GstAviDecoderClass),
+ NULL,
+ NULL,
+ (GClassInitFunc)gst_avi_decoder_class_init,
+ NULL,
+ NULL,
+ sizeof(GstAviDecoder),
+ 0,
+ (GInstanceInitFunc)gst_avi_decoder_init,
+ };
+ avi_decoder_type = g_type_register_static(GST_TYPE_BIN, "GstAviDecoder", &avi_decoder_info, 0);
+ }
+ return avi_decoder_type;
+}
+
+static void
+gst_avi_decoder_class_init (GstAviDecoderClass *klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = (GObjectClass*)klass;
+ gstelement_class = (GstElementClass*)klass;
+
+ g_object_class_install_property (G_OBJECT_CLASS(klass), ARG_BITRATE,
+ g_param_spec_long ("bitrate","bitrate","bitrate",
+ G_MINLONG, G_MAXLONG, 0, G_PARAM_READABLE)); /* CHECKME */
+ g_object_class_install_property (G_OBJECT_CLASS(klass), ARG_MEDIA_TIME,
+ g_param_spec_long ("media_time","media_time","media_time",
+ G_MINLONG, G_MAXLONG, 0, G_PARAM_READABLE)); /* CHECKME */
+ g_object_class_install_property (G_OBJECT_CLASS(klass), ARG_CURRENT_TIME,
+ g_param_spec_long ("current_time","current_time","current_time",
+ G_MINLONG, G_MAXLONG, 0, G_PARAM_READABLE)); /* CHECKME */
+
+ parent_class = g_type_class_ref (GST_TYPE_BIN);
+
+ gobject_class->get_property = gst_avi_decoder_get_property;
+}
+
+static void
+gst_avi_decoder_new_pad (GstElement *element, GstPad *pad, GstAviDecoder *avi_decoder)
+{
+ GstCaps *caps;
+ GstCaps *targetcaps = NULL;
+ const gchar *format;
+ gboolean type_found;
+ GstElement *type;
+ GstElement *new_element = NULL;
+ gchar *padname = NULL;
+ gchar *gpadname = NULL;
+#define AVI_TYPE_VIDEO 1
+#define AVI_TYPE_AUDIO 2
+ gint media_type = 0;
+
+ GST_DEBUG (0, "avidecoder: new pad for element \"%s\"\n", gst_element_get_name (element));
+
+ caps = gst_pad_get_caps (pad);
+ format = gst_caps_get_string (caps, "format");
+
+ if (!strcmp (format, "strf_vids")) {
+ targetcaps = gst_padtemplate_get_caps (GST_PADTEMPLATE_GET (src_video_templ));
+ media_type = AVI_TYPE_VIDEO;
+ gpadname = g_strdup_printf ("video_%02d", avi_decoder->video_count++);
+ }
+ else if (!strcmp (format, "strf_auds")) {
+ targetcaps = gst_padtemplate_get_caps (GST_PADTEMPLATE_GET (src_audio_templ));
+ media_type = AVI_TYPE_AUDIO;
+ gpadname = g_strdup_printf ("audio_%02d", avi_decoder->audio_count++);
+ }
+ else if (!strcmp (format, "strf_iavs")) {
+ targetcaps = gst_padtemplate_get_caps (GST_PADTEMPLATE_GET (src_video_templ));
+ media_type = AVI_TYPE_VIDEO;
+ gpadname = g_strdup_printf ("video_%02d", avi_decoder->video_count++);
+ }
+ else {
+ g_assert_not_reached ();
+ }
+
+ gst_element_set_state (GST_ELEMENT (avi_decoder), GST_STATE_PAUSED);
+
+ type = gst_elementfactory_make ("avitypes",
+ g_strdup_printf ("typeconvert%d", avi_decoder->count));
+
+ /* brin the element to the READY state so it can do our caps negotiation */
+ gst_element_set_state (type, GST_STATE_READY);
+
+ gst_pad_connect (pad, gst_element_get_pad (type, "sink"));
+ type_found = gst_util_get_bool_arg (G_OBJECT (type), "type_found");
+
+ if (type_found) {
+
+ gst_bin_add (GST_BIN (avi_decoder), type);
+
+ pad = gst_element_get_pad (type, "src");
+ caps = gst_pad_get_caps (pad);
+
+ if (gst_caps_check_compatibility (caps, targetcaps)) {
+ gst_element_add_ghost_pad (GST_ELEMENT (avi_decoder),
+ gst_element_get_pad (type, "src"), gpadname);
+
+ avi_decoder->count++;
+ goto done;
+ }
+#ifndef GST_DISABLE_AUTOPLUG
+ else {
+ GstAutoplug *autoplug;
+ autoplug = gst_autoplugfactory_make("static");
+
+ new_element = gst_autoplug_to_caps (autoplug, caps, targetcaps, NULL);
+
+ padname = "src_00";
+ }
+#endif /* GST_DISABLE_AUTOPLUG */
+ }
+
+ if (!new_element && (media_type == AVI_TYPE_VIDEO)) {
+ padname = "src";
+ }
+ else if (!new_element && (media_type == AVI_TYPE_AUDIO)) {
+ /*FIXME */
+ padname = "src";
+ }
+
+ if (new_element) {
+ gst_pad_connect (pad, gst_element_get_pad (new_element, "sink"));
+ gst_element_set_name (new_element, g_strdup_printf ("element%d", avi_decoder->count));
+ gst_bin_add (GST_BIN (avi_decoder), new_element);
+
+ gst_element_add_ghost_pad (GST_ELEMENT (avi_decoder),
+ gst_element_get_pad (new_element, padname), gpadname);
+
+ avi_decoder->count++;
+ }
+ else {
+ g_warning ("avidecoder: could not autoplug\n");
+ }
+
+done:
+ gst_element_set_state (GST_ELEMENT (avi_decoder), GST_STATE_PLAYING);
+}
+
+static void
+gst_avi_decoder_init (GstAviDecoder *avi_decoder)
+{
+ avi_decoder->demuxer = gst_elementfactory_make ("avidemux", "demux");
+
+ if (avi_decoder->demuxer) {
+ gst_bin_add (GST_BIN (avi_decoder), avi_decoder->demuxer);
+
+ gst_element_add_ghost_pad (GST_ELEMENT (avi_decoder),
+ gst_element_get_pad (avi_decoder->demuxer, "sink"), "sink");
+
+ g_signal_connect (G_OBJECT (avi_decoder->demuxer),"new_pad", G_CALLBACK (gst_avi_decoder_new_pad),
+ avi_decoder);
+ }
+ else {
+ g_warning ("wow!, no avi demuxer found. help me\n");
+ }
+
+ avi_decoder->count = 0;
+ avi_decoder->audio_count = 0;
+ avi_decoder->video_count = 0;
+}
+
+static GstCaps*
+avi_typefind (GstBuffer *buf,
+ gpointer private)
+{
+ gchar *data = GST_BUFFER_DATA (buf);
+ GstCaps *new;
+
+ GST_DEBUG (0,"avi_decoder: typefind\n");
+ if (strncmp (&data[0], "RIFF", 4)) return NULL;
+ if (strncmp (&data[8], "AVI ", 4)) return NULL;
+
+ new = GST_CAPS_NEW ("avi_typefind",
+ "video/avi",
+ "RIFF", GST_PROPS_STRING ("AVI"));
+
+ return new;
+}
+
+static void
+gst_avi_decoder_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
+{
+ GstAviDecoder *src;
+
+ g_return_if_fail (GST_IS_AVI_DECODER (object));
+
+ src = GST_AVI_DECODER (object);
+
+ switch(prop_id) {
+ case ARG_BITRATE:
+ break;
+ case ARG_MEDIA_TIME:
+ g_value_set_long (value, gst_util_get_long_arg (G_OBJECT (src->demuxer), "media_time"));
+ break;
+ case ARG_CURRENT_TIME:
+ g_value_set_long (value, gst_util_get_long_arg (G_OBJECT (src->demuxer), "current_time"));
+ break;
+ default:
+ break;
+ }
+}
+
+
+static gboolean
+plugin_init (GModule *module, GstPlugin *plugin)
+{
+ GstElementFactory *factory;
+ GstTypeFactory *type;
+
+ /* create an elementfactory for the avi_decoder element */
+ factory = gst_elementfactory_new ("avidecoder", GST_TYPE_AVI_DECODER,
+ &gst_avi_decoder_details);
+ g_return_val_if_fail (factory != NULL, FALSE);
+
+ gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (src_audio_templ));
+ gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (src_video_templ));
+ gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (sink_templ));
+
+ type = gst_typefactory_new (&avidefinition);
+ gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type));
+
+ gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
+
+ return TRUE;
+}
+
+GstPluginDesc plugin_desc = {
+ GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "avidecoder",
+ plugin_init
+};
+
avi_demux->state = GST_AVI_DEMUX_UNKNOWN;
avi_demux->num_audio_pads = 0;
avi_demux->num_video_pads = 0;
+ //avi_demux->next_time = 500000;
avi_demux->next_time = 0;
+ avi_demux->init_audio = 0;
avi_demux->flags = 0;
avi_demux->index_entries = NULL;
avi_demux->index_size = 0;
GST_INFO (GST_CAT_PLUGIN_INFO, "gst_avi_demux: samplesize %d", GUINT32_FROM_LE (strh->samplesize));
avi_demux->fcc_type = GUINT32_FROM_LE (strh->type);
+ if (strh->type == GST_RIFF_FCC_auds) {
+ guint32 scale;
+
+ scale = GUINT32_FROM_LE (strh->scale);
+ avi_demux->init_audio = GUINT32_FROM_LE (strh->init_frames);
+ if (!scale)
+ scale = 1;
+ avi_demux->audio_rate = GUINT32_FROM_LE (strh->rate) / scale;
+ }
+ else if (strh->type == GST_RIFF_FCC_vids) {
+ gfloat frame_rate;
+ guint32 scale;
+
+ scale = GUINT32_FROM_LE (strh->scale);
+ if (!scale)
+ scale = 1;
+ frame_rate = (gfloat)GUINT32_FROM_LE (strh->rate) / scale;
+
+ gst_element_send_event (GST_ELEMENT (avi_demux),
+ gst_event_new_info ("frame_rate", GST_PROPS_FLOAT (frame_rate), NULL));
+ }
return TRUE;
}
static void
gst_avidemux_parse_index (GstAviDemux *avi_demux,
- gulong offset)
+ gulong filepos, gulong offset)
{
GstBuffer *buf;
gulong index_size;
- buf = gst_pad_pullregion (avi_demux->sinkpad, GST_REGION_OFFSET_LEN, offset, 8);
+ if (!gst_bytestream_seek (avi_demux->bs, GST_SEEK_BYTEOFFSET_SET, filepos + offset)) {
+ GST_INFO (GST_CAT_PLUGIN_INFO, "avidemux: could not seek to index");
+ return;
+ }
+ buf = gst_bytestream_read (avi_demux->bs, 8);
+ while (!buf) {
+ guint32 remaining;
+ GstEvent *event;
+
+ gst_bytestream_get_status (avi_demux->bs, &remaining, &event);
- if (!buf || GST_BUFFER_OFFSET (buf) != offset || GST_BUFFER_SIZE (buf) != 8) {
+ buf = gst_bytestream_read (avi_demux->bs, 8);
+ }
+
+ if (GST_BUFFER_OFFSET (buf) != filepos + offset || GST_BUFFER_SIZE (buf) != 8) {
GST_INFO (GST_CAT_PLUGIN_INFO, "avidemux: could not get index");
return;
}
}
index_size = GUINT32_FROM_LE(*(guint32 *)(GST_BUFFER_DATA (buf) + 4));
+ gst_buffer_unref (buf);
- buf = gst_pad_pullregion(avi_demux->sinkpad, GST_REGION_OFFSET_LEN, offset+8, index_size);
+ buf = gst_bytestream_read (avi_demux->bs, index_size);
avi_demux->index_size = index_size/sizeof(gst_riff_index_entry);
avi_demux->index_entries = g_malloc (GST_BUFFER_SIZE (buf));
memcpy (avi_demux->index_entries, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+ gst_buffer_unref (buf);
- buf = gst_pad_pullregion(avi_demux->sinkpad, GST_REGION_OFFSET_LEN, avi_demux->index_offset, 0);
-}
-
-static void
-gst_avidemux_forall_pads (GstAviDemux *avi_demux, GFunc func, gpointer user_data)
-{
- gint i;
- GstPad *pad;
-
- for(i=0; i<GST_AVI_DEMUX_MAX_AUDIO_PADS; i++) {
- pad = avi_demux->audio_pad[i];
- if (pad && GST_PAD_IS_CONNECTED (pad)) {
- (*func) (pad, user_data);
- }
- }
-
- for(i=0; i<GST_AVI_DEMUX_MAX_VIDEO_PADS; i++) {
- pad = avi_demux->video_pad[i];
- if (pad && GST_PAD_IS_CONNECTED (pad)) {
- (*func) (pad, user_data);
- }
+ if (!gst_bytestream_seek (avi_demux->bs, GST_SEEK_BYTEOFFSET_SET, filepos)) {
+ GST_INFO (GST_CAT_PLUGIN_INFO, "avidemux: could not seek back to movi");
+ return;
}
}
gst_riff_chunk *chunk;
GstByteStream *bs = avi_demux->bs;
- chunk = (gst_riff_chunk *) gst_bytestream_peek_bytes (bs, sizeof (gst_riff_chunk));
- if (chunk) {
- *id = GUINT32_FROM_LE (chunk->id);
- *size = GUINT32_FROM_LE (chunk->size);
+ do {
+ chunk = (gst_riff_chunk *) gst_bytestream_peek_bytes (bs, sizeof (gst_riff_chunk));
+ if (chunk) {
+ *id = GUINT32_FROM_LE (chunk->id);
+ *size = GUINT32_FROM_LE (chunk->size);
- gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
+ gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
- return TRUE;
- }
- return gst_avidemux_handle_event (avi_demux);
+ return TRUE;
+ }
+ } while (gst_avidemux_handle_event (avi_demux));
+
+ return TRUE;
}
static gboolean
{
guint32 datashowed;
guint32 subchunksize = 0; /* size of a read subchunk */
+ gchar *formtype;
- /* flush the form type */
- if (!gst_bytestream_flush (bs, sizeof (guint32)))
+ formtype = gst_bytestream_peek_bytes (bs, sizeof (guint32));
+ if (!formtype)
return FALSE;
+ switch (GUINT32_FROM_LE (*((guint32*)formtype))) {
+ case GST_RIFF_LIST_movi:
+ gst_avidemux_parse_index (avi_demux, *filepos, *chunksize);
+ while (!gst_bytestream_flush (bs, sizeof (guint32))) {
+ guint32 remaining;
+ GstEvent *event;
+
+ gst_bytestream_get_status (avi_demux->bs, &remaining, &event);
+ }
+ break;
+ default:
+ /* flush the form type */
+ gst_bytestream_flush_fast (bs, sizeof (guint32));
+ break;
+ }
+
datashowed = sizeof (guint32); /* we showed the form type */
*filepos += datashowed; /* for the rest of the routine */
while (datashowed < *chunksize) { /* while not showed all: */
+ GST_INFO (GST_CAT_PLUGIN_INFO, "process chunk filepos %08llx", *filepos);
/* recurse for subchunks of RIFF and LIST chunks: */
if (!gst_avidemux_process_chunk (avi_demux, filepos, 0,
rec_depth + 1, &subchunksize))
subchunksize = ((subchunksize + 1) & ~1);
datashowed += (sizeof (guint32) + sizeof (guint32) + subchunksize);
- *filepos += subchunksize;
+ GST_INFO (GST_CAT_PLUGIN_INFO, "process chunk done filepos %08llx, subchunksize %08x",
+ *filepos, subchunksize);
}
if (datashowed != *chunksize) {
g_warning ("error parsing AVI");
buf = gst_bytestream_peek (bs, *chunksize);
GST_BUFFER_TIMESTAMP (buf) = avi_demux->next_time;
+
avi_demux->next_time += avi_demux->time_interval;
if (avi_demux->video_need_flush[0]) {
GST_DEBUG (0,"gst_avi_demux_chain: tag found %08x size %08x\n",
chunkid, *chunksize);
+ if (avi_demux->init_audio) {
+ //avi_demux->next_time += (*chunksize) * 1000000LL / avi_demux->audio_rate;
+ avi_demux->init_audio--;
+ }
+
if (GST_PAD_IS_CONNECTED (avi_demux->audio_pad[0])) {
GstBuffer *buf;
if (*chunksize) {
buf = gst_bytestream_peek (bs, *chunksize);
+ GST_BUFFER_TIMESTAMP (buf) = -1LL;
+
if (avi_demux->audio_need_flush[0]) {
GST_DEBUG (0,"audio flush\n");
avi_demux->audio_need_flush[0] = FALSE;
GST_INFO (GST_CAT_PLUGIN_INFO, "chunkid %s, flush %08x, filepos %08llx",
gst_riff_id_to_fourcc (chunkid), *chunksize, *filepos);
+ *filepos += *chunksize;
if (!gst_bytestream_flush (bs, *chunksize)) {
return gst_avidemux_handle_event (avi_demux);
}
gulong current_frame;
guint32 flags;
+ guint32 init_audio;
+ guint32 audio_rate;
guint num_audio_pads;
guint num_video_pads;
plugin_LTLIBRARIES = libgstossaudio.la libgstosshelper.la
-libgstossaudio_la_SOURCES = gstosssink.c gstosssrc.c gstossaudio.c gstossgst.c
+libgstossaudio_la_SOURCES = gstosssink.c gstosssrc.c gstossaudio.c gstossgst.c gstossclock.c
libgstossaudio_la_CFLAGS = $(GST_CFLAGS)
libgstossaudio_la_LIBADD = $(GST_LIBS)
libgstossaudio_la_LDFLAGS = @GST_PLUGIN_LDFLAGS@
libgstosshelper_la_SOURCES = gstosshelper.c
libgstosshelper_la_LDFLAGS = @GST_PLUGIN_LDFLAGS@
-noinst_HEADERS = gstosssink.h gstosssrc.h gstossgst.h gstosshelper.h
+noinst_HEADERS = gstosssink.h gstosssrc.h gstossgst.h gstosshelper.h gstossclock.h
static void gst_osssink_close_audio (GstOssSink *sink);
static gboolean gst_osssink_sync_parms (GstOssSink *osssink);
static GstElementStateReturn gst_osssink_change_state (GstElement *element);
+static void gst_osssink_set_clock (GstElement *element, GstClock *clock);
+static GstClock* gst_osssink_get_clock (GstElement *element);
static GstPadConnectReturn gst_osssink_sinkconnect (GstPad *pad, GstCaps *caps);
static void gst_osssink_set_property (GObject *object, guint prop_id, const GValue *value,
static void
gst_osssink_finalize (GObject *object)
{
- GstOssSink *osssink = (GstOssSink *) object;
+ GstOssSink *osssink = (GstOssSink *) object;
- g_free (osssink->device);
+ g_free (osssink->device);
- G_OBJECT_CLASS (parent_class)->finalize (object);
+ G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
osssink->device = g_strdup ("/dev/dsp");
osssink->fd = -1;
- osssink->clock = gst_clock_get_system();
osssink->channels = 1;
osssink->frequency = 11025;
osssink->fragment = 6;
#else
osssink->format = AFMT_S16_LE;
#endif /* WORDS_BIGENDIAN */
- gst_clock_register (osssink->clock, GST_OBJECT (osssink));
+ //gst_clock_register (osssink->clock, GST_OBJECT (osssink));
osssink->bufsize = 4096;
+ osssink->offset = 0LL;
/* 6 buffers per chunk by default */
osssink->sinkpool = gst_buffer_pool_get_default (osssink->bufsize, 6);
+
+ osssink->provided_clock = GST_CLOCK (gst_oss_clock_new ("OssClock", GST_ELEMENT (osssink)));
+
+ GST_ELEMENT (osssink)->setclockfunc = gst_osssink_set_clock;
+ GST_ELEMENT (osssink)->getclockfunc = gst_osssink_get_clock;
GST_FLAG_SET (osssink, GST_ELEMENT_THREAD_SUGGESTED);
}
if (width != depth)
return GST_PAD_CONNECT_REFUSED;
+ osssink->bps = 0;
+
law = gst_caps_get_int (caps, "law");
endianness = gst_caps_get_int (caps, "endianness");
sign = gst_caps_get_boolean (caps, "signed");
else if (endianness == G_BIG_ENDIAN)
format = AFMT_U16_BE;
}
+ osssink->bps = 2;
}
else if (width == 8) {
if (sign == TRUE) {
else {
format = AFMT_U8;
}
+ osssink->bps = 1;
}
}
osssink->channels = gst_caps_get_int (caps, "channels");
osssink->frequency = gst_caps_get_int (caps, "rate");
+ osssink->bps *= osssink->channels;
+ osssink->bps *= osssink->frequency;
+
if (!gst_osssink_sync_parms (osssink)) {
return GST_PAD_CONNECT_REFUSED;
}
gint target_channels;
gint target_frequency;
- g_return_if_fail (osssink != NULL);
- g_return_if_fail (GST_IS_OSSSINK (osssink));
+ g_return_val_if_fail (osssink != NULL, FALSE);
+ g_return_val_if_fail (GST_IS_OSSSINK (osssink), FALSE);
if (osssink->fd == -1)
return FALSE;
ioctl (osssink->fd, SNDCTL_DSP_CHANNELS, &osssink->channels);
ioctl (osssink->fd, SNDCTL_DSP_SPEED, &osssink->frequency);
- ioctl (osssink->fd, SNDCTL_DSP_GETBLKSIZE, &frag);
+ ioctl (osssink->fd, SNDCTL_DSP_GETBLKSIZE, &osssink->fragment);
ioctl (osssink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
GST_INFO (GST_CAT_PLUGIN_INFO, "osssink: set sound card to %dHz %d bit %s (%d bytes buffer, %08x fragment)",
osssink->frequency, osssink->format,
- (osssink->channels == 2) ? "stereo" : "mono", ospace.bytes, frag);
+ (osssink->channels == 2) ? "stereo" : "mono", ospace.bytes, osssink->fragment);
gst_element_send_event (GST_ELEMENT (osssink),
gst_event_new_info ("samplerate", GST_PROPS_INT (osssink->frequency), NULL));
gst_element_send_event (GST_ELEMENT (osssink),
gst_event_new_info ("bits", GST_PROPS_INT (osssink->format), NULL));
+ osssink->fragment_time = (1000000 * osssink->fragment) / osssink->bps;
+ GST_INFO (GST_CAT_PLUGIN_INFO, "fragment time %lu %llu\n", osssink->bps, osssink->fragment_time);
+
if (target_format != osssink->format ||
target_channels != osssink->channels ||
target_frequency != osssink->frequency)
return TRUE;
}
+static void
+gst_osssink_set_clock (GstElement *element, GstClock *clock)
+{
+ GstOssSink *osssink;
+
+ osssink = GST_OSSSINK (element);
+
+ osssink->clock = clock;
+}
+
+static GstClock*
+gst_osssink_get_clock (GstElement *element)
+{
+ GstOssSink *osssink;
+
+ osssink = GST_OSSSINK (element);
+
+ return osssink->provided_clock;
+}
+
static void
gst_osssink_chain (GstPad *pad, GstBuffer *buf)
{
GstOssSink *osssink;
- gboolean in_flush;
- audio_buf_info ospace;
+ GstClockTime buftime;
- g_return_if_fail (pad != NULL);
- g_return_if_fail (GST_IS_PAD (pad));
- g_return_if_fail (buf != NULL);
-
-
/* this has to be an audio buffer */
osssink = GST_OSSSINK (gst_pad_get_parent (pad));
-// g_return_if_fail(GST_FLAG_IS_SET(osssink,GST_STATE_RUNNING));
- if (GST_IS_EVENT (buf)) {
- gst_pad_event_default (pad, GST_EVENT (buf));
- return;
- }
+ buftime = GST_BUFFER_TIMESTAMP (buf);
+
+ if (osssink->fd >= 0) {
+ if (!osssink->mute) {
+ guchar *data = GST_BUFFER_DATA (buf);
+ gint size = GST_BUFFER_SIZE (buf);
+
+ if (osssink->clock) {
+ if (osssink->clock == osssink->provided_clock) {
+ guint64 time;
+ gint granularity, granularity_time;
+ count_info optr;
+ audio_buf_info ospace;
+ gint queued;
+
+ /* FIXME, NEW_MEDIA/DISCONT?. Try to get our start point */
+ if (osssink->offset == 0LL && buftime != -1LL) {
+ //gst_oss_clock_set_base (GST_OSS_CLOCK (osssink->clock), buftime);
+ osssink->offset = buftime;
+ }
+
+ ioctl (osssink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
+ ioctl (osssink->fd, SNDCTL_DSP_GETOPTR, &optr);
+
+ queued = (ospace.fragstotal * ospace.fragsize) - ospace.bytes;
+ time = osssink->offset + (optr.bytes) * 1000000LL / osssink->bps;
+
+ GST_DEBUG (GST_PLUGIN_INFO, "sync %llu %llu %d\n", buftime, time, queued);
+
+ granularity = ospace.fragsize;
+ //granularity = size;
+ granularity_time = granularity * osssink->fragment_time / ospace.fragsize;
+
+ while (size > 0) {
+ write (osssink->fd, data, MIN (size, granularity));
+ data += granularity;
+ size -= granularity;
+ time += granularity_time;
+ gst_clock_set_time (osssink->provided_clock, time);
+ }
+ }
+ else {
+ gst_element_clock_wait (GST_ELEMENT (osssink), osssink->clock, buftime);
+
+ write (osssink->fd, data, size);
+ }
+ }
+ else {
+ audio_buf_info ospace;
- g_signal_emit (G_OBJECT (osssink), gst_osssink_signals[SIGNAL_HANDOFF], 0,
- osssink);
-
- if (GST_BUFFER_DATA (buf) != NULL) {
-#ifndef GST_DISABLE_TRACE
- gst_trace_add_entry(NULL, 0, buf, "osssink: writing to soundcard");
-#endif // GST_DISABLE_TRACE
- //g_print("osssink: writing to soundcard\n");
- if (osssink->fd >= 0) {
- if (!osssink->mute) {
- gst_clock_wait (osssink->clock, GST_BUFFER_TIMESTAMP (buf), GST_OBJECT (osssink));
ioctl (osssink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
- GST_DEBUG (GST_CAT_PLUGIN_INFO,"osssink: (%d bytes buffer) %d %p %d\n", ospace.bytes,
- osssink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
- write (osssink->fd, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
- //write(STDOUT_FILENO,GST_BUFFER_DATA(buf),GST_BUFFER_SIZE(buf));
+
+ if (ospace.bytes >= size) {
+ write (osssink->fd, data, size);
+ }
}
}
}
case GST_STATE_READY_TO_PAUSED:
break;
case GST_STATE_PAUSED_TO_PLAYING:
+ gst_oss_clock_set_update (GST_OSS_CLOCK (osssink->provided_clock), TRUE);
break;
case GST_STATE_PLAYING_TO_PAUSED:
- if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN))
- ioctl (osssink->fd, SNDCTL_DSP_RESET, 0);
+ {
+ if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN)) {
+ if (osssink->bps) {
+ GstClockTime time;
+ audio_buf_info ospace;
+ count_info optr;
+ gint queued;
+
+ ioctl (osssink->fd, SNDCTL_DSP_GETOSPACE, &ospace);
+ ioctl (osssink->fd, SNDCTL_DSP_GETOPTR, &optr);
+
+ queued = (ospace.fragstotal * ospace.fragsize) - ospace.bytes;
+ time = (optr.bytes + queued) * 1000000LL / osssink->bps;
+ ioctl (osssink->fd, SNDCTL_DSP_RESET, 0);
+
+ gst_oss_clock_set_update (GST_OSS_CLOCK (osssink->provided_clock), FALSE);
+ gst_clock_set_time (osssink->provided_clock, time);
+ }
+ else {
+ ioctl (osssink->fd, SNDCTL_DSP_RESET, 0);
+ gst_oss_clock_set_update (GST_OSS_CLOCK (osssink->provided_clock), FALSE);
+ }
+ }
+
break;
+ }
case GST_STATE_PAUSED_TO_READY:
+ if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN))
+ ioctl (osssink->fd, SNDCTL_DSP_RESET, 0);
break;
case GST_STATE_READY_TO_NULL:
if (GST_FLAG_IS_SET (element, GST_OSSSINK_OPEN))
#include <config.h>
#include <gst/gst.h>
+#include "gstossclock.h"
#ifdef __cplusplus
extern "C" {
GstPad *sinkpad;
GstBufferPool *sinkpool;
- //GstClockTime clocktime;
+ GstClock *provided_clock;
GstClock *clock;
/* device */
gint fragment;
gboolean mute;
guint bufsize;
+ guint bps;
+ guint64 offset;
+
+ guint64 fragment_time;
};
struct _GstOssSinkClass {
projects / platform / upstream / gst-plugins-good.git / commitdiff