Vitor Sessak a écrit :
> On 07/13/2010 10:01 PM, Sebastian Vater wrote:
>> Vitor Sessak a écrit :
>>> On 07/11/2010 10:04 PM, Sebastian Vater wrote:
>>>> Vitor Sessak a écrit :
>>>>> On 07/07/2010 10:46 PM, Sebastian Vater wrote:
>>>>>>
>>>>>> diff --git a/libavsequencer/song.h b/libavsequencer/song.h
>>>> /*
>>>> * AVSequencer sub-song management
>>>> * Copyright (c) 2010 Sebastian
>>>> Vater<cdgs.basty-gM/[email protected]>
>>>> *
>>>> * This file is part of FFmpeg.
>>>> *
>>>> * FFmpeg is free software; you can redistribute it and/or
>>>> * modify it under the terms of the GNU Lesser General Public
>>>> * License as published by the Free Software Foundation; either
>>>> * version 2.1 of the License, or (at your option) any later version.
>>>> *
>>>> * FFmpeg 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
>>>> * Lesser General Public License for more details.
>>>> *
>>>> * You should have received a copy of the GNU Lesser General Public
>>>> * License along with FFmpeg; if not, write to the Free Software
>>>> * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
>>>> 02110-1301 USA
>>>> */
>>>>
>>>> #ifndef AVSEQUENCER_SONG_H
>>>> #define AVSEQUENCER_SONG_H
>>>>
>>>> #include "libavformat/avformat.h"
>>>> #include "libavsequencer/avsequencer.h"
>>>> #include "libavsequencer/order.h"
>>>> #include "libavsequencer/track.h"
>>>> #include "libavsequencer/player.h"
>>>>
>>>> /**
>>>> * Sequencer song structure.
>>>> * New fields can be added to the end with minor version bumps.
>>>> * Removal, reordering and changes to existing fields require a major
>>>> * version bump.
>>>> */
>>>> typedef struct AVSequencerSong {
>>>> /** Metadata information: Original sub-song file name, sub-song
>>>> * title, song message, artist, genre, album, begin and finish
>>>> * date of composition and comment. */
>>>> AVMetadata *metadata;
>>>
>>>> /** AVSequencerPlayerGlobals pointer to global channel data. */
>>>> AVSequencerPlayerGlobals *global_data;
>>>>
>>>> /** AVSequencerPlayerHostChannel pointer to host channel
>>>> data. */
>>>> AVSequencerPlayerHostChannel *channel_data;
>>>
>>> Player?
>>
>> Can be moved to AVSequencerPlayerGlobals but would make some of the
>> playback code requiring more accesses, but I will fix this!
>>
>>>
>>>
>>>> /** AVSequencerOrderList pointer to list of order data. */
>>>> AVSequencerOrderList *order_list;
>>>
>>> Why not here an array of pointers like everywhere else?
>>
>> There is only one order_list per sub-song, but multiple order list
>> entries.
>>
>>>
>>>> /** Array of pointers containing all tracks for this
>>>> sub-song. */
>>>> AVSequencerTrack **track_list;
>>>>
>>>> /** Duration of this sub-song, in AV_TIME_BASE fractional
>>>> seconds. */
>>>> uint64_t duration;
>>>>
>>>> /** Number of tracks attached to this sub-song. */
>>>> uint16_t tracks;
>>>
>>>> /** Stack size, i.e. maximum recursion depth of GoSub command
>>>> which
>>>> defaults to 4. */
>>>> uint16_t gosub_stack_size;
>>>> #define AVSEQ_SONG_GOSUB_STACK 4
>>>
>>> Doesn't this depend on the player implementation? Or is it
>>> format-specific? Or is it read from the file?
>>
>> GOSUB is a TuComposer only feature right now. I thought this to be a
>> nice default value, for creating a new sub-song.
>
> Ok, but does the player need to know this value or it could just fail
> if the song use more recursion depth than the player support?
>
>>>> /** Stack size, i.e. maximum recursion depth of the pattern loop
>>>> command, which defaults to 1 to imitate most trackers. */
>>>> uint16_t loop_stack_size;
>>>> #define AVSEQ_SONG_PATTERN_LOOP_STACK 1
>>>
>>> Again, is this specified in the file?
>>
>> This is, however, is the standard (and even only possible value) for all
>> non-TuComposer trackers, no trackers otherwise allow nesting of the
>> pattern loop command.
>
> same.
>
>>>> /** Compatibility flags for playback. There are rare cases
>>>> where effect handling can not be mapped into internal
>>>> playback engine and have to be handled specially. For
>>>> each sub-song which needs this, this will define new
>>>> flags which tag the player to handle it to that special
>>>> way. */
>>>> uint8_t compat_flags;
>>>> #define AVSEQ_SONG_COMPAT_FLAG_SYNC 0x01 ///< Tracks are
>>>> synchronous (linked together, pattern based)
>>>> #define AVSEQ_SONG_COMPAT_FLAG_GLOBAL_LOOP 0x02 ///< Global
>>>> pattern loop memory
>>>> #define AVSEQ_SONG_COMPAT_FLAG_AMIGA_LIMITS 0x04 ///< Enforce
>>>> AMIGA sound hardware limits (portamento)
>>>> #define AVSEQ_SONG_COMPAT_FLAG_OLD_VOLUMES 0x08 ///< All volume
>>>> related commands range from 0-64 instead of 0-255
>>>> #define AVSEQ_SONG_COMPAT_FLAG_GLOBAL_NEW_ONLY 0x10 ///< Global
>>>> volume/panning changes affect new notes only (S3M)
>>>>
>>>> /** Song playback flags. Some sequencers use a totally
>>>> different timing scheme which has to be taken care
>>>> specially in the internal playback engine. Also
>>>> sequencers differ in how they handle slides. */
>>>> uint8_t flags;
>>>> #define AVSEQ_SONG_FLAG_LINEAR_FREQ_TABLE 0x01 ///< Use linear
>>>> instead of Amiga frequency table
>>>> #define AVSEQ_SONG_FLAG_SPD 0x02 ///< Use SPD
>>>> (OctaMED style) timing instead of BpM
>>>> #define AVSEQ_SONG_FLAG_MONO 0x04 ///< Use mono
>>>> instead of stereo output
>>>> #define AVSEQ_SONG_FLAG_SURROUND 0x08 ///< Initial global
>>>> surround instead of stereo panning
>>>>
>>>> /** Maximum number of host channels, as edited in the track view.
>>>> to be allocated and usable for order list (defaults to
>>>> 16). */
>>>> uint16_t channels;
>>>> #define AVSEQ_SONG_CHANNELS 16
>>>> #define AVSEQ_SONG_CHANNELS_MIN 1
>>>> #define AVSEQ_SONG_CHANNELS_MAX 256
>>>>
>>>> /** Initial number of frames per row, i.e. sequencer tempo
>>>> (defaults to 6 as in most tracker formats). */
>>>> uint16_t frames;
>>>> #define AVSEQ_SONG_FRAMES 6
>>>>
>>>> /** Initial speed multiplier, i.e. nominator which defaults
>>>> to disabled = 0. */
>>>> uint8_t speed_mul;
>>>> #define AVSEQ_SONG_SPEED_MUL 0
>>>>
>>>> /** Initial speed divider, i.e. denominator which defaults
>>>> to disabled = 0. */
>>>> uint8_t speed_div;
>>>> #define AVSEQ_SONG_SPEED_DIV 0
>>>
>>> AVRational?
>>
>> Would be an idea, though. But would make the track effect manipulating
>> with this incompatible and conversions required. So for sake of
>> simplicity, I kept them all the same.
>>
>>
>>>
>>>> /** Initial MED style SPD speed (defaults to 33 as in
>>>> OctaMED Soundstudio). */
>>>> uint16_t spd_speed;
>>>> #define AVSEQ_SONG_SPD_SPEED 33
>>>>
>>>> /** Initial number of rows per beat (defaults to 4 rows are a
>>>> beat). */
>>>> uint16_t bpm_tempo;
>>>> #define AVSEQ_SONG_BPM_TEMPO 4
>>>>
>>>> /** Initial beats per minute speed (defaults to 50 Hz => 125
>>>> BpM). */
>>>> uint16_t bpm_speed;
>>>> #define AVSEQ_SONG_BPM_SPEED 125
>>>
>>>> /** Minimum and lower limit of number of frames per row
>>>> (defaults to 1). */
>>>> uint16_t frames_min;
>>>> #define AVSEQ_SONG_FRAMES_MIN 1
>>>
>>> Again, does this limit depend of the file format or is read directly
>>> from the file?
>>
>> Frames (=tempo) 0 is allocated for mark song end, which does not make
>> sense as a start tempo in sub-songs, which would mean that the whole
>> sub-song would initialized with song end at the very beginning.
>>
>> Allowing tempo 0 is like division by zero, it does not make sense,
>> because otherwise infinite rows and tracks would be processed in just
>> one tick.
>
> Yes, so why is this a variable not a define? What happens with the
> player if a file specifies frames_min == 2 and this field is
> erroneously initialized frames_min == 1?
>
>>>
>>>> /** Maximum and upper limit of number of frames per row
>>>> (defaults to 255). */
>>>> uint16_t frames_max;
>>>> #define AVSEQ_SONG_FRAMES_MAX 255
>>>>
>>>> /** Minimum and lower limit of MED style SPD timing values
>>>> (defaults to 1). */
>>>> uint16_t spd_min;
>>>> #define AVSEQ_SONG_SPD_MIN 1
>>>>
>>>> /** Maximum and upper limit of MED style SPD timing values
>>>> (defaults to 255). */
>>>> uint16_t spd_max;
>>>> #define AVSEQ_SONG_SPD_MAX 255
>>>>
>>>> /** Minimum and lower limit of rows per beat timing values
>>>> (defaults to 1). */
>>>> uint16_t bpm_tempo_min;
>>>> #define AVSEQ_SONG_BPM_TEMPO_MIN 1
>>>>
>>>> /** Maximum and upper limit of rows per beat timing values
>>>> (defaults to 255). */
>>>> uint16_t bpm_tempo_max;
>>>> #define AVSEQ_SONG_BPM_TEMPO_MAX 255
>>>>
>>>> /** Minimum and lower limit of beats per minute timing values
>>>> (defaults to 1). */
>>>> uint16_t bpm_speed_min;
>>>> #define AVSEQ_SONG_BPM_SPEED_MIN 1
>>>>
>>>> /** Maximum and upper limit of beats per minute timing values
>>>> (defaults to 255). */
>>>> uint16_t bpm_speed_max;
>>>> #define AVSEQ_SONG_BPM_SPEED_MAX 255
>>>
>>> Same for those.
>>
>> Would overflow multiply in playback time calculation for larger
>> values...and also does not make sense, see track effects, they all set
>> 8-bit values only, if we would allow a higher speed here, we could never
>> change the speed values which are larger than 255.
>
> same.
>
>>>
>>>> /** Global volume of this sub-song. All other volume related
>>>> commands are scaled by this (defaults to 255 = no
>>>> scaling). */
>>>> uint8_t global_volume;
>>>> #define AVSEQ_SONG_VOLUME 255
>>>>
>>>> /** Global sub-volume of this sub-song. This is basically
>>>> volume divided by 256, but the sub-volume doesn't account
>>>> into actual mixer output (defaults to 0). */
>>>> uint8_t global_sub_volume;
>>>> #define AVSEQ_SONG_SUB_VOLUME 0
>>>>
>>>> /** Global panning of this sub-song. All other panning related
>>>> commands are scaled by this stereo separation factor
>>>> (defaults to 0 which means full stereo separation). */
>>>> uint8_t global_panning;
>>>> #define AVSEQ_SONG_PANNING 0
>>>>
>>>> /** Global sub-panning of this sub-song. This is basically
>>>> panning divided by 256, but the sub-panning doesn't account
>>>> into actual mixer output (defaults to 0). */
>>>> uint8_t global_sub_panning;
>>>> #define AVSEQ_SONG_SUB_PANNING 0
>>>
>>> Again, a file can specify those, no? Or they are parameters for the
>>> player?
>>
>> Yes, although TuComposer itself is the only one now who does support
>> this, after extending from 8-bit to 16-bit I have had to find an usage
>> for the lower 8-bits of the data word of track effects, these allow
>> setting it.
>>
>> But remember again, all stuff in
>> module.h/song.h/order.h/track.h/instr.h/sample.h/synth.h is supposed to
>> be changed by editors, too. Only player.h is mostly read-only for
>> external applications.
>
> I don't understand this point. What can an editor do that cannot be
> read from a file? In which struct should I set the player parameters
> that are independent of the song (i.e, some global volume to scale the
> song global volume, or if the player should output floats or s16le or
> s32le)? It does not belongs to the BSS anyway...
Hi I have excellent news!
libavsequencer now flawlessly integrates into FFmpeg, just check out my
latest git. Please do a git pull --rebase, Stefano had problems without
using it.
Here are the song.[ch] part of the BSS to review.
This version compiles perfectly.
--
Best regards,
:-) Basty/CDGS (-:
diff --git a/libavsequencer/song.c b/libavsequencer/song.c
new file mode 100644
index 0000000..d767b0d
--- /dev/null
+++ b/libavsequencer/song.c
@@ -0,0 +1,91 @@
+/*
+ * Implement AVSequencer sub-song stuff
+ * Copyright (c) 2010 Sebastian Vater <[email protected]>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Implement AVSequencer sub-song stuff.
+ */
+
+#include "libavutil/log.h"
+#include "libavsequencer/avsequencer.h"
+
+static const char *song_name(void *p)
+{
+ AVSequencerSong *song = p;
+ AVMetadataTag *tag = av_metadata_get(song->metadata, "title", NULL, AV_METADATA_IGNORE_SUFFIX);
+
+ return tag->value;
+}
+
+static const AVClass avseq_song_class = {
+ "AVSequencer Song",
+ song_name,
+ NULL,
+ LIBAVUTIL_VERSION_INT,
+};
+
+int avseq_song_open(AVSequencerModule *module, AVSequencerSong *song) {
+ AVSequencerSong **song_list;
+ uint16_t songs;
+ int res;
+
+ if (!module)
+ return AVERROR_INVALIDDATA;
+
+ song_list = module->song_list;
+ songs = module->songs;
+
+ if (!(song && ++songs)) {
+ return AVERROR_INVALIDDATA;
+ } else if (!(song_list = av_realloc(song_list, songs * sizeof(AVSequencerSong *)))) {
+ av_log(module, AV_LOG_ERROR, "cannot allocate sub-song storage container.\n");
+ return AVERROR(ENOMEM);
+ }
+
+ song->av_class = &avseq_song_class;
+ song->channels = 16;
+ song->gosub_stack_size = 4;
+ song->loop_stack_size = 1;
+ song->frames = 6;
+ song->spd_speed = 33;
+ song->bpm_tempo = 4;
+ song->bpm_speed = 125;
+ song->frames_min = 1;
+ song->frames_max = 65535;
+ song->spd_min = 1;
+ song->spd_max = 65535;
+ song->bpm_tempo_min = 1;
+ song->bpm_tempo_max = 65535;
+ song->bpm_speed_min = 1;
+ song->bpm_speed_max = 65535;
+ song->global_volume = 255;
+
+ if ((res = avseq_order_open(song)) < 0) {
+ av_free(song_list);
+ return res;
+ }
+
+ song_list[songs] = song;
+ module->song_list = song_list;
+ module->songs = songs;
+
+ return 0;
+}
diff --git a/libavsequencer/song.h b/libavsequencer/song.h
new file mode 100755
index 0000000..15f8b3e
--- /dev/null
+++ b/libavsequencer/song.h
@@ -0,0 +1,208 @@
+/*
+ * AVSequencer sub-song management
+ * Copyright (c) 2010 Sebastian Vater <[email protected]>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVSEQUENCER_SONG_H
+#define AVSEQUENCER_SONG_H
+
+#include "libavutil/log.h"
+#include "libavformat/avformat.h"
+#include "libavsequencer/order.h"
+#include "libavsequencer/track.h"
+
+/** AVSequencerSong->compat_flags bitfield. */
+enum AVSequencerSongCompatFlags {
+ AVSEQ_SONG_COMPAT_FLAG_SYNC = 0x01, ///< Tracks are synchronous (linked together, pattern based)
+ AVSEQ_SONG_COMPAT_FLAG_GLOBAL_LOOP = 0x02, ///< Global pattern loop memory
+ AVSEQ_SONG_COMPAT_FLAG_AMIGA_LIMITS = 0x04, ///< Enforce AMIGA sound hardware limits (portamento)
+ AVSEQ_SONG_COMPAT_FLAG_OLD_VOLUMES = 0x08, ///< All volume related commands range from 0-64 instead of 0-255
+ AVSEQ_SONG_COMPAT_FLAG_GLOBAL_NEW_ONLY = 0x10, ///< Global volume/panning changes affect new notes only (S3M)
+};
+
+/** AVSequencerSong->compat_flags bitfield. */
+enum AVSequencerSongFlags {
+ AVSEQ_SONG_FLAG_LINEAR_FREQ_TABLE = 0x01, ///< Use linear instead of Amiga frequency table
+ AVSEQ_SONG_FLAG_SPD = 0x02, ///< Use SPD (OctaMED style) timing instead of BpM
+ AVSEQ_SONG_FLAG_MONO = 0x04, ///< Use mono instead of stereo output
+ AVSEQ_SONG_FLAG_SURROUND = 0x08, ///< Initial global surround instead of stereo panning
+};
+
+/**
+ * Sequencer song structure.
+ * New fields can be added to the end with minor version bumps.
+ * Removal, reordering and changes to existing fields require a major
+ * version bump.
+ */
+typedef struct AVSequencerSong {
+ /**
+ * information on struct for av_log
+ * - set by avseq_alloc_context
+ */
+ const AVClass *av_class;
+
+ /** Metadata information: Original sub-song file name, sub-song
+ * title, song message, artist, genre, album, begin and finish
+ * date of composition and comment. */
+ AVMetadata *metadata;
+
+ /** AVSequencerOrderList pointer to list of order data. */
+ AVSequencerOrderList *order_list;
+
+ /** Array (of size tracks) of pointers containing all tracks for
+ this sub-song. */
+ AVSequencerTrack **track_list;
+
+ /** Number of tracks attached to this sub-song. */
+ uint16_t tracks;
+
+ /** Duration of this sub-song, in AV_TIME_BASE fractional
+ seconds. */
+ uint64_t duration;
+
+ /** Stack size, i.e. maximum recursion depth of GoSub command which
+ defaults to 4. */
+ uint16_t gosub_stack_size;
+
+ /** Stack size, i.e. maximum recursion depth of the pattern loop
+ command, which defaults to 1 to imitate most trackers (most
+ trackers do not even support any other value than one, i.e.
+ the pattern loop command is not nestable). */
+ uint16_t loop_stack_size;
+
+ /** Compatibility flags for playback. There are rare cases
+ where effect handling can not be mapped into internal
+ playback engine and have to be handled specially. For
+ each sub-song which needs this, this will define new
+ flags which tag the player to handle it to that special
+ way. */
+ uint8_t compat_flags;
+
+ /** Song playback flags. Some sequencers use a totally
+ different timing scheme which has to be taken care
+ specially in the internal playback engine. Also
+ sequencers differ in how they handle slides. */
+ uint8_t flags;
+
+ /** Maximum number of host channels, as edited in the track view.
+ to be allocated and usable for order list (defaults to 16). */
+ uint16_t channels;
+
+ /** Initial number of frames per row, i.e. sequencer tempo
+ (defaults to 6 as in most tracker formats), a value of
+ zero is pointless, since that would mean to play unlimited
+ rows and tracks in just one tick. */
+ uint16_t frames;
+
+ /** Initial speed multiplier, i.e. nominator which defaults
+ to disabled = 0. */
+ uint8_t speed_mul;
+
+ /** Initial speed divider, i.e. denominator which defaults
+ to disabled = 0. */
+ uint8_t speed_div;
+
+ /** Initial MED style SPD speed (defaults to 33 as in
+ OctaMED Soundstudio). */
+ uint16_t spd_speed;
+
+ /** Initial number of rows per beat (defaults to 4 rows are a beat). */
+ uint16_t bpm_tempo;
+
+ /** Initial beats per minute speed (defaults to 50 Hz => 125 BpM). */
+ uint16_t bpm_speed;
+
+ /** Minimum and lower limit of number of frames per row
+ (defaults to 1), a value of zero is pointless, since
+ that would mean to play unlimited rows and tracks in
+ just one tick. */
+ uint16_t frames_min;
+
+ /** Maximum and upper limit of number of frames per row
+ (defaults to 255) since a larger value would not make
+ sense (see track effects, they all set 8-bit values only),
+ if we would allow a higher speed here, we could never
+ change the speed values which are larger than 255. */
+ uint16_t frames_max;
+
+ /** Minimum and lower limit of MED style SPD timing values
+ (defaults to 1). */
+ uint16_t spd_min;
+
+ /** Maximum and upper limit of MED style SPD timing values
+ (defaults to 255) since a larger value would not make
+ sense (see track effects, they all set 8-bit values only),
+ if we would allow a higher speed here, we could never
+ change the speed values which are larger than 255. */
+ uint16_t spd_max;
+
+ /** Minimum and lower limit of rows per beat timing values
+ (defaults to 1). */
+ uint16_t bpm_tempo_min;
+
+ /** Maximum and upper limit of rows per beat timing values
+ (defaults to 255) since a larger value would not make
+ sense (see track effects, they all set 8-bit values only),
+ if we would allow a higher speed here, we could never
+ change the speed values which are larger than 255. */
+ uint16_t bpm_tempo_max;
+
+ /** Minimum and lower limit of beats per minute timing values
+ (defaults to 1). */
+ uint16_t bpm_speed_min;
+
+ /** Maximum and upper limit of beats per minute timing values
+ (defaults to 255) since a larger value would not make
+ sense (see track effects, they all set 8-bit values only),
+ if we would allow a higher speed here, we could never
+ change the speed values which are larger than 255. */
+ uint16_t bpm_speed_max;
+
+ /** Global volume of this sub-song. All other volume related
+ commands are scaled by this (defaults to 255 = no scaling). */
+ uint8_t global_volume;
+
+ /** Global sub-volume of this sub-song. This is basically
+ volume divided by 256, but the sub-volume doesn't account
+ into actual mixer output (defaults to 0). */
+ uint8_t global_sub_volume;
+
+ /** Global panning of this sub-song. All other panning related
+ commands are scaled by this stereo separation factor
+ (defaults to 0 which means full stereo separation). */
+ uint8_t global_panning;
+
+ /** Global sub-panning of this sub-song. This is basically
+ panning divided by 256, but the sub-panning doesn't account
+ into actual mixer output (defaults to 0). */
+ uint8_t global_sub_panning;
+
+ /** Array of pointers containing every unknown data field where
+ the last element is indicated by a NULL pointer reference. The
+ first 64-bit of the unknown data contains an unique identifier
+ for this chunk and the second 64-bit data is actual unsigned
+ length of the following raw data. Some formats are chunk based
+ and can store information, which can't be handled by some
+ other, in case of a transition the unknown data is kept as is.
+ Some programs write editor settings for sub-songs in those
+ chunks, which then won't get lost in that case. */
+ uint8_t **unknown_data;
+} AVSequencerSong;
+
+#endif /* AVSEQUENCER_SONG_H */
_______________________________________________
FFmpeg-soc mailing list
[email protected]
https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-soc
