gstreamer/gst-libs/gst/riff/riff-read.c

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/* GStreamer RIFF I/O
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* riff-read.c: RIFF input file parsing
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] Original commit message from CVS: reviewed by David Schleef * configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] * gst-libs/gst/riff/riff-read.c: Include gst/gstutils.h. (gst_riff_peek_head, gst_riff_peek_list, gst_riff_read_list) (gst_riff_read_header): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/typefind/gsttypefindfunctions.c: Include gst/gstutils.h. (mp3_type_find): Use GST_READ_UINT* macros to access possibly unaligned memory. (mp3_type_find, mpeg1_parse_header, qt_type_find) (speex_type_find): Likewise * gst/tags/gstvorbistag.c: (ADVANCE): Likewise * gst/qtdemux/qtdemux.c: Include stdlib.h (needed by realloc). (QTDEMUX_GUINT32_GET, QTDEMUX_GUINT16_GET, QTDEMUX_FP32_GET) (QTDEMUX_FP16_GET, QTDEMUX_FOURCC_GET) (gst_qtdemux_loop_header, gst_qtdemux_loop_header) (qtdemux_node_dump_foreach, qtdemux_tree_get_child_by_type) (qtdemux_tree_get_sibling_by_type): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk): Likewise. * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead) (gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise. * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain): Likewise. * gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA) (gst_mpeg2subt_chain_subtitle): Likewise. * gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq) (gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain): Likewise. * gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info): Likewise. * gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes): Likewise. * gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint): Likewise.
2004-04-20 21:04:22 +00:00
#include <gst/gstutils.h>
#include "riff-ids.h"
#include "riff-read.h"
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
GST_DEBUG_CATEGORY_STATIC (riffread_debug);
#define GST_CAT_DEFAULT riffread_debug
enum
{
ARG_0,
ARG_METADATA
/* FILL ME */
};
static void gst_riff_read_class_init (GstRiffReadClass * klass);
static void gst_riff_read_init (GstRiffRead * riff);
static GstElementStateReturn gst_riff_read_change_state (GstElement * element);
static GstElementClass *parent_class = NULL;
GType
gst_riff_read_get_type (void)
{
static GType gst_riff_read_type = 0;
if (!gst_riff_read_type) {
static const GTypeInfo gst_riff_read_info = {
sizeof (GstRiffReadClass),
NULL,
NULL,
(GClassInitFunc) gst_riff_read_class_init,
NULL,
NULL,
sizeof (GstRiffRead),
0,
(GInstanceInitFunc) gst_riff_read_init,
};
gst_riff_read_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstRiffRead",
&gst_riff_read_info, 0);
}
return gst_riff_read_type;
}
static void
gst_riff_read_class_init (GstRiffReadClass * klass)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
GST_DEBUG_CATEGORY_INIT (riffread_debug, "riffread",
0, "RIFF stream helper class");
gstelement_class->change_state = gst_riff_read_change_state;
}
static void
gst_riff_read_init (GstRiffRead * riff)
{
riff->sinkpad = NULL;
riff->bs = NULL;
riff->level = NULL;
}
static GstElementStateReturn
gst_riff_read_change_state (GstElement * element)
{
GstRiffRead *riff = GST_RIFF_READ (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_READY_TO_PAUSED:
if (!riff->sinkpad)
return GST_STATE_FAILURE;
riff->bs = gst_bytestream_new (riff->sinkpad);
break;
case GST_STATE_PAUSED_TO_READY:
gst_bytestream_destroy (riff->bs);
while (riff->level) {
GstRiffLevel *level = riff->level->data;
riff->level = g_list_remove (riff->level, level);
g_free (level);
}
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
/*
* Return: the amount of levels in the hierarchy that the
* current element lies higher than the previous one.
* The opposite isn't done - that's auto-done using list
* element reading.
*/
static guint
gst_riff_read_element_level_up (GstRiffRead * riff)
{
guint num = 0;
guint64 pos = gst_bytestream_tell (riff->bs);
while (riff->level != NULL) {
GList *last = g_list_last (riff->level);
GstRiffLevel *level = last->data;
if (pos >= level->start + level->length) {
riff->level = g_list_remove (riff->level, level);
g_free (level);
num++;
} else
break;
}
return num;
}
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
/*
* Event handler. Basic:
* - EOS: end-of-file, stop processing, forward EOS.
* - Interrupt: stop processing.
* - Discont: shouldn't be handled here but in the seek handler. Error.
* - Flush: ignore, since we check for flush flags manually. Don't forward.
* - Others: warn, ignore.
* Return value indicates whether to continue processing.
*/
static gboolean
gst_riff_read_use_event (GstRiffRead * riff, GstEvent * event)
{
if (!event) {
GST_ELEMENT_ERROR (riff, RESOURCE, READ, (NULL), (NULL));
return FALSE;
}
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
gst_pad_event_default (riff->sinkpad, event);
return FALSE;
case GST_EVENT_INTERRUPT:
gst_event_unref (event);
return FALSE;
case GST_EVENT_DISCONTINUOUS:
GST_WARNING_OBJECT (riff, "Unexpected discont - might lose sync");
gst_event_unref (event);
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
return TRUE;
case GST_EVENT_FLUSH:
gst_event_unref (event);
return TRUE;
default:
GST_WARNING ("don't know how to handle event %d", GST_EVENT_TYPE (event));
gst_pad_event_default (riff->sinkpad, event);
return TRUE;
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
}
/* happy */
g_assert_not_reached ();
return FALSE;
}
static gboolean
gst_riff_read_handle_event (GstRiffRead * riff)
{
GstEvent *event = NULL;
guint32 remaining;
gst_bytestream_get_status (riff->bs, &remaining, &event);
return gst_riff_read_use_event (riff, event);
}
/*
* Read the next tag plus length (may be NULL). Return
* TRUE on success or FALSE on failure.
*/
gboolean
gst_riff_peek_head (GstRiffRead * riff,
guint32 * tag, guint32 * length, guint * level_up)
{
GList *last;
guint8 *data;
/* if we're at the end of a chunk, but unaligned, then re-align.
* Those are essentially broken files, but unfortunately they
* exist. */
if ((last = g_list_last (riff->level)) != NULL) {
GstRiffLevel *level = last->data;
guint64 pos = gst_bytestream_tell (riff->bs);
if (level->start + level->length - pos < 8) {
if (!gst_bytestream_flush (riff->bs, level->start + level->length - pos)) {
GST_ELEMENT_ERROR (riff, RESOURCE, READ, (NULL), (NULL));
return FALSE;
}
}
}
/* read */
while (gst_bytestream_peek_bytes (riff->bs, &data, 8) != 8) {
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
if (!gst_riff_read_handle_event (riff))
return FALSE;
}
/* parse tag + length (if wanted) */
configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] Original commit message from CVS: reviewed by David Schleef * configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] * gst-libs/gst/riff/riff-read.c: Include gst/gstutils.h. (gst_riff_peek_head, gst_riff_peek_list, gst_riff_read_list) (gst_riff_read_header): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/typefind/gsttypefindfunctions.c: Include gst/gstutils.h. (mp3_type_find): Use GST_READ_UINT* macros to access possibly unaligned memory. (mp3_type_find, mpeg1_parse_header, qt_type_find) (speex_type_find): Likewise * gst/tags/gstvorbistag.c: (ADVANCE): Likewise * gst/qtdemux/qtdemux.c: Include stdlib.h (needed by realloc). (QTDEMUX_GUINT32_GET, QTDEMUX_GUINT16_GET, QTDEMUX_FP32_GET) (QTDEMUX_FP16_GET, QTDEMUX_FOURCC_GET) (gst_qtdemux_loop_header, gst_qtdemux_loop_header) (qtdemux_node_dump_foreach, qtdemux_tree_get_child_by_type) (qtdemux_tree_get_sibling_by_type): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk): Likewise. * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead) (gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise. * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain): Likewise. * gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA) (gst_mpeg2subt_chain_subtitle): Likewise. * gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq) (gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain): Likewise. * gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info): Likewise. * gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes): Likewise. * gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint): Likewise.
2004-04-20 21:04:22 +00:00
*tag = GST_READ_UINT32_LE (data);
if (length)
configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] Original commit message from CVS: reviewed by David Schleef * configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] * gst-libs/gst/riff/riff-read.c: Include gst/gstutils.h. (gst_riff_peek_head, gst_riff_peek_list, gst_riff_read_list) (gst_riff_read_header): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/typefind/gsttypefindfunctions.c: Include gst/gstutils.h. (mp3_type_find): Use GST_READ_UINT* macros to access possibly unaligned memory. (mp3_type_find, mpeg1_parse_header, qt_type_find) (speex_type_find): Likewise * gst/tags/gstvorbistag.c: (ADVANCE): Likewise * gst/qtdemux/qtdemux.c: Include stdlib.h (needed by realloc). (QTDEMUX_GUINT32_GET, QTDEMUX_GUINT16_GET, QTDEMUX_FP32_GET) (QTDEMUX_FP16_GET, QTDEMUX_FOURCC_GET) (gst_qtdemux_loop_header, gst_qtdemux_loop_header) (qtdemux_node_dump_foreach, qtdemux_tree_get_child_by_type) (qtdemux_tree_get_sibling_by_type): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk): Likewise. * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead) (gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise. * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain): Likewise. * gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA) (gst_mpeg2subt_chain_subtitle): Likewise. * gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq) (gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain): Likewise. * gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info): Likewise. * gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes): Likewise. * gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint): Likewise.
2004-04-20 21:04:22 +00:00
*length = GST_READ_UINT32_LE (((guint32 *) data) + 1);
/* level */
if (level_up)
*level_up = gst_riff_read_element_level_up (riff);
return TRUE;
}
/*
* Read: the actual data (plus alignment and flush).
* Return: the data, as a GstBuffer.
*/
GstBuffer *
gst_riff_peek_element_data (GstRiffRead * riff, guint length, guint * got_bytes)
{
GstBuffer *buf = NULL;
guint32 got;
while ((got = gst_bytestream_peek (riff->bs, &buf, length)) != length) {
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
if (buf)
gst_buffer_unref (buf);
if (!gst_riff_read_handle_event (riff))
return NULL;
}
if (got_bytes)
*got_bytes = got;
return buf;
}
GstBuffer *
gst_riff_read_element_data (GstRiffRead * riff, guint length, guint * got_bytes)
{
GstBuffer *buf;
if (!(buf = gst_riff_peek_element_data (riff, length, got_bytes)))
return NULL;
/* we need 16-bit alignment */
if (length & 1)
length++;
if (!gst_bytestream_flush (riff->bs, length)) {
gst_buffer_unref (buf);
return NULL;
}
return buf;
}
/*
* Seek.
*/
GstEvent *
gst_riff_read_seek (GstRiffRead * riff, guint64 offset)
{
guint64 length = gst_bytestream_length (riff->bs);
guint32 remaining;
GstEvent *event = NULL;
guchar *data;
/* hack for AVI files with broken idx1 size chunk markers */
if (offset > length)
offset = length;
/* first, flush remaining buffers */
gst_bytestream_get_status (riff->bs, &remaining, &event);
if (event) {
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
GST_WARNING ("Unexpected event before seek");
if (!gst_riff_read_use_event (riff, event))
return NULL;
ext/dvdread/dvdreadsrc.c: Fix. Don't do one big huge loop around the whole DVD, that will cache all data and thus eat... Original commit message from CVS: * ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init), (dvdreadsrc_init), (dvdreadsrc_dispose), (dvdreadsrc_set_property), (dvdreadsrc_get_property), (_open), (_seek), (_read), (dvdreadsrc_get), (dvdreadsrc_open_file), (dvdreadsrc_change_state): Fix. Don't do one big huge loop around the whole DVD, that will cache all data and thus eat sizeof(dvd) (several GB) before we see something. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek): Actually NULL'ify event after using it. * gst/matroska/ebml-read.c: (gst_ebml_read_use_event), (gst_ebml_read_handle_event), (gst_ebml_read_element_id), (gst_ebml_read_element_length), (gst_ebml_read_element_data), (gst_ebml_read_seek), (gst_ebml_read_skip): Handle events. * gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init), (gst_dvd_demux_init), (gst_dvd_demux_get_audio_stream), (gst_dvd_demux_get_subpicture_stream), (gst_dvd_demux_plugin_init): Fix timing (this will probably break if I seek using menus, but I didn't get there yet). VOBs and normal DVDs should now work. Add a mpeg2-only pad with high rank so this get autoplugged for MPEG-2 movies. * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_base_init), (gst_mpeg_demux_class_init), (gst_mpeg_demux_init), (gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_get_video_stream), (gst_mpeg_demux_get_audio_stream), (gst_mpeg_demux_get_private_stream), (gst_mpeg_demux_parse_packet), (gst_mpeg_demux_parse_pes), (gst_mpeg_demux_plugin_init): Use this as second rank for MPEG-1 and MPEG-2. Still use this for MPEG-1 but use dvddemux for MPEG-2. * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init), (gst_mpeg_parse_init), (gst_mpeg_parse_new_pad), (gst_mpeg_parse_parse_packhead): Timing. Only add pad template if it exists. Add sink template from class and not from ourselves. This means we will always use the correct sink template even if it is not the one defined in this file.
2004-10-01 08:42:56 +00:00
event = NULL;
}
if (remaining)
gst_bytestream_flush_fast (riff->bs, remaining);
/* now seek */
if (!gst_bytestream_seek (riff->bs, offset, GST_SEEK_METHOD_SET)) {
GST_ELEMENT_ERROR (riff, RESOURCE, SEEK, (NULL), (NULL));
return NULL;
}
/* and now, peek a new byte. This will fail because there's a
* pending event. Then, take the event and return it. */
while (!event) {
if (gst_bytestream_peek_bytes (riff->bs, &data, 1)) {
GST_WARNING ("Unexpected data after seek - this means seek failed");
break;
}
/* get the discont event and return */
gst_bytestream_get_status (riff->bs, &remaining, &event);
if (!event) {
GST_WARNING ("No discontinuity event after seek - seek failed");
break;
} else if (GST_EVENT_TYPE (event) != GST_EVENT_DISCONTINUOUS) {
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
if (!gst_riff_read_use_event (riff, event))
return NULL;
event = NULL;
}
}
return event;
}
/*
* Gives the tag of the next RIFF element.
*/
guint32
gst_riff_peek_tag (GstRiffRead * riff, guint * level_up)
{
guint32 tag;
if (!gst_riff_peek_head (riff, &tag, NULL, level_up))
return 0;
return tag;
}
/*
* Gives the tag of the next LIST/RIFF element.
*/
guint32
gst_riff_peek_list (GstRiffRead * riff)
{
guint32 lst;
guint8 *data;
if (!gst_riff_peek_head (riff, &lst, NULL, NULL))
return FALSE;
if (lst != GST_RIFF_TAG_LIST) {
g_warning ("Not a LIST object");
return 0;
}
if (gst_bytestream_peek_bytes (riff->bs, &data, 12) != 12) {
GST_ELEMENT_ERROR (riff, RESOURCE, READ, (NULL), (NULL));
return 0;
}
configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] Original commit message from CVS: reviewed by David Schleef * configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] * gst-libs/gst/riff/riff-read.c: Include gst/gstutils.h. (gst_riff_peek_head, gst_riff_peek_list, gst_riff_read_list) (gst_riff_read_header): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/typefind/gsttypefindfunctions.c: Include gst/gstutils.h. (mp3_type_find): Use GST_READ_UINT* macros to access possibly unaligned memory. (mp3_type_find, mpeg1_parse_header, qt_type_find) (speex_type_find): Likewise * gst/tags/gstvorbistag.c: (ADVANCE): Likewise * gst/qtdemux/qtdemux.c: Include stdlib.h (needed by realloc). (QTDEMUX_GUINT32_GET, QTDEMUX_GUINT16_GET, QTDEMUX_FP32_GET) (QTDEMUX_FP16_GET, QTDEMUX_FOURCC_GET) (gst_qtdemux_loop_header, gst_qtdemux_loop_header) (qtdemux_node_dump_foreach, qtdemux_tree_get_child_by_type) (qtdemux_tree_get_sibling_by_type): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk): Likewise. * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead) (gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise. * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain): Likewise. * gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA) (gst_mpeg2subt_chain_subtitle): Likewise. * gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq) (gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain): Likewise. * gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info): Likewise. * gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes): Likewise. * gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint): Likewise.
2004-04-20 21:04:22 +00:00
return GST_READ_UINT32_LE (((guint32 *) data) + 2);
}
/*
* Don't read data.
*/
gboolean
gst_riff_read_skip (GstRiffRead * riff)
{
guint32 tag, length;
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
GstEvent *event = NULL;
guint32 remaining;
if (!gst_riff_peek_head (riff, &tag, &length, NULL))
return FALSE;
/* 16-bit alignment */
if (length & 1)
length++;
/* header itself */
length += 8;
/* see if we have that much data available */
gst_bytestream_get_status (riff->bs, &remaining, &event);
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
if (event) {
GST_WARNING ("Unexpected event in skip");
if (!gst_riff_read_use_event (riff, event))
return FALSE;
}
/* yes */
if (remaining >= length) {
gst_bytestream_flush_fast (riff->bs, length);
return TRUE;
}
/* no */
if (!(event = gst_riff_read_seek (riff,
gst_bytestream_tell (riff->bs) + length)))
return FALSE;
gst_event_unref (event);
return TRUE;
}
/*
* Read any type of data.
*/
gboolean
gst_riff_read_data (GstRiffRead * riff, guint32 * tag, GstBuffer ** buf)
{
guint32 length;
if (!gst_riff_peek_head (riff, tag, &length, NULL))
return FALSE;
gst_bytestream_flush_fast (riff->bs, 8);
return ((*buf = gst_riff_read_element_data (riff, length, NULL)) != NULL);
}
/*
* Read a string.
*/
gboolean
gst_riff_read_ascii (GstRiffRead * riff, guint32 * tag, gchar ** str)
{
GstBuffer *buf;
if (!gst_riff_read_data (riff, tag, &buf))
return FALSE;
*str = g_malloc (GST_BUFFER_SIZE (buf) + 1);
memcpy (*str, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
(*str)[GST_BUFFER_SIZE (buf)] = '\0';
gst_buffer_unref (buf);
return TRUE;
}
/*
* Read media structs.
*/
gboolean
gst_riff_read_strh (GstRiffRead * riff, gst_riff_strh ** header)
{
guint32 tag;
GstBuffer *buf;
gst_riff_strh *strh;
if (!gst_riff_read_data (riff, &tag, &buf))
return FALSE;
if (tag != GST_RIFF_TAG_strh) {
g_warning ("Not a strh chunk");
gst_buffer_unref (buf);
return FALSE;
}
if (GST_BUFFER_SIZE (buf) < sizeof (gst_riff_strh)) {
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
GST_WARNING ("Too small strh (%d available, %d needed)",
GST_BUFFER_SIZE (buf), (int) sizeof (gst_riff_strh));
gst_buffer_unref (buf);
return FALSE;
}
strh = g_memdup (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
gst_buffer_unref (buf);
#if (G_BYTE_ORDER == G_BIG_ENDIAN)
strh->type = GUINT32_FROM_LE (strh->type);
strh->fcc_handler = GUINT32_FROM_LE (strh->fcc_handler);
strh->flags = GUINT32_FROM_LE (strh->flags);
strh->priority = GUINT32_FROM_LE (strh->priority);
strh->init_frames = GUINT32_FROM_LE (strh->init_frames);
strh->scale = GUINT32_FROM_LE (strh->scale);
strh->rate = GUINT32_FROM_LE (strh->rate);
strh->start = GUINT32_FROM_LE (strh->start);
strh->length = GUINT32_FROM_LE (strh->length);
strh->bufsize = GUINT32_FROM_LE (strh->bufsize);
strh->quality = GUINT32_FROM_LE (strh->quality);
strh->samplesize = GUINT32_FROM_LE (strh->samplesize);
#endif
/* avoid divisions by zero */
if (!strh->scale)
strh->scale = 1;
if (!strh->rate)
strh->rate = 1;
/* debug */
GST_INFO ("strh tag found");
GST_INFO (" type " GST_FOURCC_FORMAT, GST_FOURCC_ARGS (strh->type));
GST_INFO (" fcc_handler " GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (strh->fcc_handler));
GST_INFO (" flags 0x%08x", strh->flags);
GST_INFO (" priority %d", strh->priority);
GST_INFO (" init_frames %d", strh->init_frames);
GST_INFO (" scale %d", strh->scale);
GST_INFO (" rate %d", strh->rate);
GST_INFO (" start %d", strh->start);
GST_INFO (" length %d", strh->length);
GST_INFO (" bufsize %d", strh->bufsize);
GST_INFO (" quality %d", strh->quality);
GST_INFO (" samplesize %d", strh->samplesize);
*header = strh;
return TRUE;
}
gboolean
gst/: Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. Thi... Original commit message from CVS: reviewed by: <delete if not using a buddy> * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps_with_data), (gst_riff_create_video_caps), (gst_riff_create_audio_caps), (gst_riff_create_video_template_caps), (gst_riff_create_audio_template_caps): * gst-libs/gst/riff/riff-media.h: * gst-libs/gst/riff/riff-read.c: (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids): * gst-libs/gst/riff/riff-read.h: * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream): Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. This is usually color palettes (as in RLE, but also in 8-bit RGB). Also use those during caps creation. Lastly, add ADPCM (similar to wavparse - which should eventually be rifflib based). * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init), (gst_matroska_demux_init), (gst_matroska_demux_reset): * gst/matroska/matroska-demux.h: Remove placeholders for some prehistoric tagging system. Didn't add support for any tag system really anyway. * gst/qtdemux/qtdemux.c: Add support for audio/x-m4a (MPEG-4) through spider. * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt), (gst_wavparse_loop): ADPCM support (#135862). Increase max. buffer size because we cannot split buffers for ADPCM (screws references) and I've seen files with 2048 byte chunks. 4096 seems safe for now.
2004-04-16 01:20:44 +00:00
gst_riff_read_strf_vids_with_data (GstRiffRead * riff,
gst_riff_strf_vids ** header, GstBuffer ** extradata)
{
guint32 tag;
GstBuffer *buf;
gst_riff_strf_vids *strf;
if (!gst_riff_read_data (riff, &tag, &buf))
return FALSE;
if (tag != GST_RIFF_TAG_strf) {
g_warning ("Not a strf chunk");
gst_buffer_unref (buf);
return FALSE;
}
if (GST_BUFFER_SIZE (buf) < sizeof (gst_riff_strf_vids)) {
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
GST_WARNING ("Too small strf_vids (%d available, %d needed)",
GST_BUFFER_SIZE (buf), (int) sizeof (gst_riff_strf_vids));
gst_buffer_unref (buf);
return FALSE;
}
strf = g_memdup (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
#if (G_BYTE_ORDER == G_BIG_ENDIAN)
strf->size = GUINT32_FROM_LE (strf->size);
strf->width = GUINT32_FROM_LE (strf->width);
strf->height = GUINT32_FROM_LE (strf->height);
strf->planes = GUINT16_FROM_LE (strf->planes);
strf->bit_cnt = GUINT16_FROM_LE (strf->bit_cnt);
strf->compression = GUINT32_FROM_LE (strf->compression);
strf->image_size = GUINT32_FROM_LE (strf->image_size);
strf->xpels_meter = GUINT32_FROM_LE (strf->xpels_meter);
strf->ypels_meter = GUINT32_FROM_LE (strf->ypels_meter);
strf->num_colors = GUINT32_FROM_LE (strf->num_colors);
strf->imp_colors = GUINT32_FROM_LE (strf->imp_colors);
#endif
/* size checking */
gst/: Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. Thi... Original commit message from CVS: reviewed by: <delete if not using a buddy> * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps_with_data), (gst_riff_create_video_caps), (gst_riff_create_audio_caps), (gst_riff_create_video_template_caps), (gst_riff_create_audio_template_caps): * gst-libs/gst/riff/riff-media.h: * gst-libs/gst/riff/riff-read.c: (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids): * gst-libs/gst/riff/riff-read.h: * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream): Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. This is usually color palettes (as in RLE, but also in 8-bit RGB). Also use those during caps creation. Lastly, add ADPCM (similar to wavparse - which should eventually be rifflib based). * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init), (gst_matroska_demux_init), (gst_matroska_demux_reset): * gst/matroska/matroska-demux.h: Remove placeholders for some prehistoric tagging system. Didn't add support for any tag system really anyway. * gst/qtdemux/qtdemux.c: Add support for audio/x-m4a (MPEG-4) through spider. * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt), (gst_wavparse_loop): ADPCM support (#135862). Increase max. buffer size because we cannot split buffers for ADPCM (screws references) and I've seen files with 2048 byte chunks. 4096 seems safe for now.
2004-04-16 01:20:44 +00:00
*extradata = NULL;
if (strf->size > GST_BUFFER_SIZE (buf)) {
g_warning ("strf_vids header gave %d bytes data, only %d available",
strf->size, GST_BUFFER_SIZE (buf));
strf->size = GST_BUFFER_SIZE (buf);
gst/: Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. Thi... Original commit message from CVS: reviewed by: <delete if not using a buddy> * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps_with_data), (gst_riff_create_video_caps), (gst_riff_create_audio_caps), (gst_riff_create_video_template_caps), (gst_riff_create_audio_template_caps): * gst-libs/gst/riff/riff-media.h: * gst-libs/gst/riff/riff-read.c: (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids): * gst-libs/gst/riff/riff-read.h: * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream): Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. This is usually color palettes (as in RLE, but also in 8-bit RGB). Also use those during caps creation. Lastly, add ADPCM (similar to wavparse - which should eventually be rifflib based). * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init), (gst_matroska_demux_init), (gst_matroska_demux_reset): * gst/matroska/matroska-demux.h: Remove placeholders for some prehistoric tagging system. Didn't add support for any tag system really anyway. * gst/qtdemux/qtdemux.c: Add support for audio/x-m4a (MPEG-4) through spider. * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt), (gst_wavparse_loop): ADPCM support (#135862). Increase max. buffer size because we cannot split buffers for ADPCM (screws references) and I've seen files with 2048 byte chunks. 4096 seems safe for now.
2004-04-16 01:20:44 +00:00
} else if (strf->size < GST_BUFFER_SIZE (buf)) {
gint len;
len = GST_BUFFER_SIZE (buf) - strf->size;
if (len > 0) {
*extradata = gst_buffer_create_sub (buf, strf->size, len);
}
} else if (strf->size > sizeof (gst_riff_strf_vids)) {
*extradata = gst_buffer_create_sub (buf,
sizeof (gst_riff_strf_vids), strf->size - sizeof (gst_riff_strf_vids));
}
/* debug */
GST_INFO ("strf tag found in context vids:");
GST_INFO (" size %d", strf->size);
GST_INFO (" width %d", strf->width);
GST_INFO (" height %d", strf->height);
GST_INFO (" planes %d", strf->planes);
GST_INFO (" bit_cnt %d", strf->bit_cnt);
GST_INFO (" compression " GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (strf->compression));
GST_INFO (" image_size %d", strf->image_size);
GST_INFO (" xpels_meter %d", strf->xpels_meter);
GST_INFO (" ypels_meter %d", strf->ypels_meter);
GST_INFO (" num_colors %d", strf->num_colors);
GST_INFO (" imp_colors %d", strf->imp_colors);
gst/: Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. Thi... Original commit message from CVS: reviewed by: <delete if not using a buddy> * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps_with_data), (gst_riff_create_video_caps), (gst_riff_create_audio_caps), (gst_riff_create_video_template_caps), (gst_riff_create_audio_template_caps): * gst-libs/gst/riff/riff-media.h: * gst-libs/gst/riff/riff-read.c: (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids): * gst-libs/gst/riff/riff-read.h: * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream): Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. This is usually color palettes (as in RLE, but also in 8-bit RGB). Also use those during caps creation. Lastly, add ADPCM (similar to wavparse - which should eventually be rifflib based). * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init), (gst_matroska_demux_init), (gst_matroska_demux_reset): * gst/matroska/matroska-demux.h: Remove placeholders for some prehistoric tagging system. Didn't add support for any tag system really anyway. * gst/qtdemux/qtdemux.c: Add support for audio/x-m4a (MPEG-4) through spider. * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt), (gst_wavparse_loop): ADPCM support (#135862). Increase max. buffer size because we cannot split buffers for ADPCM (screws references) and I've seen files with 2048 byte chunks. 4096 seems safe for now.
2004-04-16 01:20:44 +00:00
if (*extradata)
GST_INFO (" %d bytes extra_data", GST_BUFFER_SIZE (*extradata));
gst_buffer_unref (buf);
*header = strf;
return TRUE;
}
gst/: Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. Thi... Original commit message from CVS: reviewed by: <delete if not using a buddy> * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps_with_data), (gst_riff_create_video_caps), (gst_riff_create_audio_caps), (gst_riff_create_video_template_caps), (gst_riff_create_audio_template_caps): * gst-libs/gst/riff/riff-media.h: * gst-libs/gst/riff/riff-read.c: (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids): * gst-libs/gst/riff/riff-read.h: * gst/avi/gstavidemux.c: (gst_avi_demux_add_stream): Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. This is usually color palettes (as in RLE, but also in 8-bit RGB). Also use those during caps creation. Lastly, add ADPCM (similar to wavparse - which should eventually be rifflib based). * gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init), (gst_matroska_demux_init), (gst_matroska_demux_reset): * gst/matroska/matroska-demux.h: Remove placeholders for some prehistoric tagging system. Didn't add support for any tag system really anyway. * gst/qtdemux/qtdemux.c: Add support for audio/x-m4a (MPEG-4) through spider. * gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt), (gst_wavparse_loop): ADPCM support (#135862). Increase max. buffer size because we cannot split buffers for ADPCM (screws references) and I've seen files with 2048 byte chunks. 4096 seems safe for now.
2004-04-16 01:20:44 +00:00
/*
* Obsolete, use gst_riff_read_strf_vids_with_data ().
*/
gboolean
gst_riff_read_strf_vids (GstRiffRead * riff, gst_riff_strf_vids ** header)
{
GstBuffer *data = NULL;
gboolean ret;
ret = gst_riff_read_strf_vids_with_data (riff, header, &data);
if (data)
gst_buffer_unref (data);
return ret;
}
gboolean
gst_riff_read_strf_auds_with_data (GstRiffRead * riff,
gst_riff_strf_auds ** header, GstBuffer ** extradata)
{
guint32 tag;
GstBuffer *buf;
gst_riff_strf_auds *strf;
if (!gst_riff_read_data (riff, &tag, &buf))
return FALSE;
if (tag != GST_RIFF_TAG_strf && tag != GST_RIFF_TAG_fmt) {
g_warning ("Not a strf chunk");
gst_buffer_unref (buf);
return FALSE;
}
if (GST_BUFFER_SIZE (buf) < sizeof (gst_riff_strf_auds)) {
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
GST_WARNING ("Too small strf_auds (%d available, %d needed)",
GST_BUFFER_SIZE (buf), (int) sizeof (gst_riff_strf_auds));
gst_buffer_unref (buf);
return FALSE;
}
strf = g_memdup (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
#if (G_BYTE_ORDER == G_BIG_ENDIAN)
strf->format = GUINT16_FROM_LE (strf->format);
strf->channels = GUINT16_FROM_LE (strf->channels);
strf->rate = GUINT32_FROM_LE (strf->rate);
strf->av_bps = GUINT32_FROM_LE (strf->av_bps);
strf->blockalign = GUINT16_FROM_LE (strf->blockalign);
strf->size = GUINT16_FROM_LE (strf->size);
#endif
/* size checking */
*extradata = NULL;
if (GST_BUFFER_SIZE (buf) > sizeof (gst_riff_strf_auds) + 2) {
gint len;
len = GST_READ_UINT16_LE (&GST_BUFFER_DATA (buf)[16]);
if (len + 2 + sizeof (gst_riff_strf_auds) > GST_BUFFER_SIZE (buf)) {
GST_WARNING ("Extradata indicated %d bytes, but only %d available",
len, GST_BUFFER_SIZE (buf) - 2 - sizeof (gst_riff_strf_auds));
len = GST_BUFFER_SIZE (buf) - 2 - sizeof (gst_riff_strf_auds);
}
if (len > 0) {
*extradata = gst_buffer_create_sub (buf,
sizeof (gst_riff_strf_auds) + 2, len);
}
}
/* debug */
GST_INFO ("strf tag found in context auds:");
GST_INFO (" format %d", strf->format);
GST_INFO (" channels %d", strf->channels);
GST_INFO (" rate %d", strf->rate);
GST_INFO (" av_bps %d", strf->av_bps);
GST_INFO (" blockalign %d", strf->blockalign);
GST_INFO (" size %d", strf->size); /* wordsize, not extrasize! */
if (*extradata)
GST_INFO (" %d bytes extra_data", GST_BUFFER_SIZE (*extradata));
gst_buffer_unref (buf);
*header = strf;
return TRUE;
}
/*
* Obsolete, use gst_riff_read_strf_auds_with_data ().
*/
gboolean
gst_riff_read_strf_auds (GstRiffRead * riff, gst_riff_strf_auds ** header)
{
GstBuffer *data = NULL;
gboolean ret;
ret = gst_riff_read_strf_auds_with_data (riff, header, &data);
if (data)
gst_buffer_unref (data);
return ret;
}
gboolean
gst_riff_read_strf_iavs (GstRiffRead * riff, gst_riff_strf_iavs ** header)
{
guint32 tag;
GstBuffer *buf;
gst_riff_strf_iavs *strf;
if (!gst_riff_read_data (riff, &tag, &buf))
return FALSE;
if (tag != GST_RIFF_TAG_strf) {
g_warning ("Not a strf chunk");
gst_buffer_unref (buf);
return FALSE;
}
if (GST_BUFFER_SIZE (buf) < sizeof (gst_riff_strf_iavs)) {
ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing... Original commit message from CVS: * ext/flac/gstflacdec.c: (gst_flacdec_src_query): Only return true if we actually filled something in. Prevents player applications from showing a random length for flac files. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init), (gst_riff_read_use_event), (gst_riff_read_handle_event), (gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh), (gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs): OK, ok, so I implemented event handling. Apparently it's normal that we receive random events at random points without asking for it. * gst/avi/gstavidemux.c: (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index), (gst_avi_demux_sync), (gst_avi_demux_stream_scan), (gst_avi_demux_massage_index), (gst_avi_demux_stream_header), (gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data), (gst_avi_demux_loop): * gst/avi/gstavidemux.h: Implement non-lineair chunk handling and subchunk processing. The first solves playback of AVI files where the audio and video data of individual buffers that we read are not synchronized. This should not happen according to the wonderful AVI specs, but of course it does happen in reality. It is also a prerequisite for the second. Subchunk processing allows us to cut chunks in small pieces and process each of these pieces separately. This is required because I've seen several AVI files with incredibly large audio chunks, even some files with only one audio chunk for the whole file. This allows for proper playback including seeking. This patch is supposed to fix all AVI A/V sync issues. * gst/flx/gstflxdec.c: (gst_flxdec_class_init), (flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop): Work. * gst/modplug/gstmodplug.cc: Proper return value setting for the query() function. * gst/playback/gstplaybasebin.c: (setup_source): Being in non-playing state (after, e.g., EOS) is not necessarily a bad thing. Allow for that. This fixes playback of short files. They don't actually playback fully now, because the clock already runs. This means that small files (<500kB) with a small length (<2sec) will still not or barely play. Other files, such as mod or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
GST_WARNING ("Too small strf_iavs (%d available, %d needed)",
GST_BUFFER_SIZE (buf), (int) sizeof (gst_riff_strf_iavs));
gst_buffer_unref (buf);
return FALSE;
}
strf = g_memdup (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
gst_buffer_unref (buf);
#if (G_BYTE_ORDER == G_BIG_ENDIAN)
strf->DVAAuxSrc = GUINT32_FROM_LE (strf->DVAAuxSrc);
strf->DVAAuxCtl = GUINT32_FROM_LE (strf->DVAAuxCtl);
strf->DVAAuxSrc1 = GUINT32_FROM_LE (strf->DVAAuxSrc1);
strf->DVAAuxCtl1 = GUINT32_FROM_LE (strf->DVAAuxCtl1);
strf->DVVAuxSrc = GUINT32_FROM_LE (strf->DVVAuxSrc);
strf->DVVAuxCtl = GUINT32_FROM_LE (strf->DVVAuxCtl);
strf->DVReserved1 = GUINT32_FROM_LE (strf->DVReserved1);
strf->DVReserved2 = GUINT32_FROM_LE (strf->DVReserved2);
#endif
/* debug */
GST_INFO ("strf tag found in context iavs");
GST_INFO (" DVAAuxSrc %08x", strf->DVAAuxSrc);
GST_INFO (" DVAAuxCtl %08x", strf->DVAAuxCtl);
GST_INFO (" DVAAuxSrc1 %08x", strf->DVAAuxSrc1);
GST_INFO (" DVAAuxCtl1 %08x", strf->DVAAuxCtl1);
GST_INFO (" DVVAuxSrc %08x", strf->DVVAuxSrc);
GST_INFO (" DVVAuxCtl %08x", strf->DVVAuxCtl);
GST_INFO (" DVReserved1 %08x", strf->DVReserved1);
GST_INFO (" DVReserved2 %08x", strf->DVReserved2);
*header = strf;
return TRUE;
}
/*
* Read a list.
*/
gboolean
gst_riff_read_list (GstRiffRead * riff, guint32 * tag)
{
guint32 length, lst;
GstRiffLevel *level;
guint8 *data;
if (!gst_riff_peek_head (riff, &lst, &length, NULL))
return FALSE;
if (lst != GST_RIFF_TAG_LIST) {
g_warning ("Not a LIST object");
return FALSE;
}
gst_bytestream_flush_fast (riff->bs, 8);
if (gst_bytestream_peek_bytes (riff->bs, &data, 4) != 4) {
GST_ELEMENT_ERROR (riff, RESOURCE, READ, (NULL), (NULL));
return FALSE;
}
gst_bytestream_flush_fast (riff->bs, 4);
configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] Original commit message from CVS: reviewed by David Schleef * configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] * gst-libs/gst/riff/riff-read.c: Include gst/gstutils.h. (gst_riff_peek_head, gst_riff_peek_list, gst_riff_read_list) (gst_riff_read_header): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/typefind/gsttypefindfunctions.c: Include gst/gstutils.h. (mp3_type_find): Use GST_READ_UINT* macros to access possibly unaligned memory. (mp3_type_find, mpeg1_parse_header, qt_type_find) (speex_type_find): Likewise * gst/tags/gstvorbistag.c: (ADVANCE): Likewise * gst/qtdemux/qtdemux.c: Include stdlib.h (needed by realloc). (QTDEMUX_GUINT32_GET, QTDEMUX_GUINT16_GET, QTDEMUX_FP32_GET) (QTDEMUX_FP16_GET, QTDEMUX_FOURCC_GET) (gst_qtdemux_loop_header, gst_qtdemux_loop_header) (qtdemux_node_dump_foreach, qtdemux_tree_get_child_by_type) (qtdemux_tree_get_sibling_by_type): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk): Likewise. * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead) (gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise. * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain): Likewise. * gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA) (gst_mpeg2subt_chain_subtitle): Likewise. * gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq) (gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain): Likewise. * gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info): Likewise. * gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes): Likewise. * gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint): Likewise.
2004-04-20 21:04:22 +00:00
*tag = GST_READ_UINT32_LE (data);
/* remember level */
level = g_new (GstRiffLevel, 1);
level->start = gst_bytestream_tell (riff->bs);
level->length = length - 4;
riff->level = g_list_append (riff->level, level);
return TRUE;
}
/*
* Utility function for reading metadata in a RIFF file.
*/
gboolean
gst_riff_read_info (GstRiffRead * riff)
{
guint32 tag;
guint64 end;
GstRiffLevel *level;
GList *last;
gchar *name, *type;
GstTagList *taglist;
gboolean have_tags = FALSE;
/* What we're doing here is ugly (oh no!); we look
* at our LIST tag size and assure that we do not
* cross boundaries. This is to maintain the level
* counter for the client app. */
last = g_list_last (riff->level);
level = last->data;
riff->level = g_list_remove (riff->level, level);
end = level->start + level->length;
g_free (level);
taglist = gst_tag_list_new ();
while (gst_bytestream_tell (riff->bs) < end) {
if (!gst_riff_peek_head (riff, &tag, NULL, NULL)) {
return FALSE;
}
/* find out the type of metadata */
switch (tag) {
case GST_RIFF_INFO_IARL:
type = GST_TAG_LOCATION;
break;
case GST_RIFF_INFO_IART:
type = GST_TAG_ARTIST;
break;
case GST_RIFF_INFO_ICMS:
type = NULL; /*"Commissioner"; */
break;
case GST_RIFF_INFO_ICMT:
type = GST_TAG_COMMENT;
break;
case GST_RIFF_INFO_ICOP:
type = GST_TAG_COPYRIGHT;
break;
case GST_RIFF_INFO_ICRD:
type = GST_TAG_DATE;
break;
case GST_RIFF_INFO_ICRP:
type = NULL; /*"Cropped"; */
break;
case GST_RIFF_INFO_IDIM:
type = NULL; /*"Dimensions"; */
break;
case GST_RIFF_INFO_IDPI:
type = NULL; /*"Dots per Inch"; */
break;
case GST_RIFF_INFO_IENG:
type = NULL; /*"Engineer"; */
break;
case GST_RIFF_INFO_IGNR:
type = GST_TAG_GENRE;
break;
case GST_RIFF_INFO_IKEY:
type = NULL; /*"Keywords"; */ ;
break;
case GST_RIFF_INFO_ILGT:
type = NULL; /*"Lightness"; */
break;
case GST_RIFF_INFO_IMED:
type = NULL; /*"Medium"; */
break;
case GST_RIFF_INFO_INAM:
type = GST_TAG_TITLE;
break;
case GST_RIFF_INFO_IPLT:
type = NULL; /*"Palette"; */
break;
case GST_RIFF_INFO_IPRD:
type = NULL; /*"Product"; */
break;
case GST_RIFF_INFO_ISBJ:
type = NULL; /*"Subject"; */
break;
case GST_RIFF_INFO_ISFT:
type = GST_TAG_ENCODER;
break;
case GST_RIFF_INFO_ISHP:
type = NULL; /*"Sharpness"; */
break;
case GST_RIFF_INFO_ISRC:
type = GST_TAG_ISRC;
break;
case GST_RIFF_INFO_ISRF:
type = NULL; /*"Source Form"; */
break;
case GST_RIFF_INFO_ITCH:
type = NULL; /*"Technician"; */
break;
default:
type = NULL;
GST_WARNING ("Unknown INFO (metadata) tag entry " GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (tag));
break;
}
if (type) {
name = NULL;
if (!gst_riff_read_ascii (riff, &tag, &name)) {
return FALSE;
}
if (name && name[0] != '\0') {
GValue src = { 0 }
ext/dirac/: Do something. Don't actually know if this works because I don't have a demuxer yet. Original commit message from CVS: * ext/dirac/Makefile.am: * ext/dirac/gstdirac.cc: * ext/dirac/gstdiracdec.cc: * ext/dirac/gstdiracdec.h: Do something. Don't actually know if this works because I don't have a demuxer yet. * ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps): Add channels=1 to caps returned from _getcaps(). * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type), (gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_parse_init), (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init), (gst_ogm_parse_sink_convert), (gst_ogm_parse_chain), (gst_ogm_parse_change_state): Separate between audio/video so ogmaudioparse actually uses the audio pad templates. Both audio and video work now, including autoplugging. Also use sometimes-srcpad hack. * gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek): Handle events better. Don't hang on infinite loops. * gst/avi/gstavidemux.c: (gst_avi_demux_class_init), (gst_avi_demux_init), (gst_avi_demux_reset), (gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query), (gst_avi_demux_stream_header), (gst_avi_demux_stream_data), (gst_avi_demux_change_state): * gst/avi/gstavidemux.h: Improve A/V sync. Still not perfect. * gst/matroska/ebml-read.c: (gst_ebml_read_seek), (gst_ebml_read_skip): Handle events better. * gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event), (gst_qtdemux_loop_header), (qtdemux_parse_trak), (qtdemux_audio_caps): Add IMA4. Improve event handling. Save offset after a seek when the headers are at the end of the file so that we don't end up in an infinite loop. * gst/typefind/gsttypefindfunctions.c: (qt_type_find): Add low-priority typefind support for files with no length.
2004-09-23 14:59:22 +00:00
, dest = {
0};
GType dest_type = gst_tag_get_type (type);
have_tags = TRUE;
g_value_init (&src, G_TYPE_STRING);
g_value_set_string (&src, name);
g_value_init (&dest, dest_type);
g_value_transform (&src, &dest);
g_value_unset (&src);
gst_tag_list_add_values (taglist, GST_TAG_MERGE_APPEND,
type, &dest, NULL);
g_value_unset (&dest);
}
g_free (name);
} else {
gst_riff_read_skip (riff);
}
}
if (have_tags) {
GstElement *element = GST_ELEMENT (riff);
GstEvent *event = gst_event_new_tag (taglist);
const GList *padlist;
/* let the world know about this wonderful thing */
for (padlist = gst_element_get_pad_list (element);
padlist != NULL; padlist = padlist->next) {
if (GST_PAD_IS_SRC (padlist->data) && GST_PAD_IS_USABLE (padlist->data)) {
gst_event_ref (event);
gst_pad_push (GST_PAD (padlist->data), GST_DATA (event));
}
}
gst_element_found_tags (element, taglist);
gst_event_unref (event);
} else {
gst_tag_list_free (taglist);
}
return TRUE;
}
/*
* Read RIFF header and document type.
*/
gboolean
gst_riff_read_header (GstRiffRead * riff, guint32 * doctype)
{
GstRiffLevel *level;
guint32 tag, length;
guint8 *data;
/* We ignore size for openDML-2.0 support */
if (!gst_riff_peek_head (riff, &tag, &length, NULL))
return FALSE;
if (tag != GST_RIFF_TAG_RIFF) {
GST_ELEMENT_ERROR (riff, STREAM, WRONG_TYPE, (NULL), (NULL));
return FALSE;
}
gst_bytestream_flush_fast (riff->bs, 8);
/* doctype */
if (gst_bytestream_peek_bytes (riff->bs, &data, 4) != 4) {
GST_ELEMENT_ERROR (riff, RESOURCE, READ, (NULL), (NULL));
return FALSE;
}
gst_bytestream_flush_fast (riff->bs, 4);
configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] Original commit message from CVS: reviewed by David Schleef * configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds] * gst-libs/gst/riff/riff-read.c: Include gst/gstutils.h. (gst_riff_peek_head, gst_riff_peek_list, gst_riff_read_list) (gst_riff_read_header): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/typefind/gsttypefindfunctions.c: Include gst/gstutils.h. (mp3_type_find): Use GST_READ_UINT* macros to access possibly unaligned memory. (mp3_type_find, mpeg1_parse_header, qt_type_find) (speex_type_find): Likewise * gst/tags/gstvorbistag.c: (ADVANCE): Likewise * gst/qtdemux/qtdemux.c: Include stdlib.h (needed by realloc). (QTDEMUX_GUINT32_GET, QTDEMUX_GUINT16_GET, QTDEMUX_FP32_GET) (QTDEMUX_FP16_GET, QTDEMUX_FOURCC_GET) (gst_qtdemux_loop_header, gst_qtdemux_loop_header) (qtdemux_node_dump_foreach, qtdemux_tree_get_child_by_type) (qtdemux_tree_get_sibling_by_type): Use GST_READ_UINT* macros to access possibly unaligned memory. * gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk): Likewise. * gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead) (gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise. * gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain): Likewise. * gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA) (gst_mpeg2subt_chain_subtitle): Likewise. * gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq) (gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain): Likewise. * gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info): Likewise. * gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes): Likewise. * gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint): Likewise.
2004-04-20 21:04:22 +00:00
*doctype = GST_READ_UINT32_LE (data);
/* remember level */
level = g_new (GstRiffLevel, 1);
level->start = gst_bytestream_tell (riff->bs);
level->length = length - 4;
riff->level = g_list_append (riff->level, level);
return TRUE;
}