mpv/demux/demux.c

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/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* mpv 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <assert.h>
#include <unistd.h>
#include <pthread.h>
#include <math.h>
#include <sys/types.h>
#include <sys/stat.h>
#include "config.h"
#include "options/options.h"
#include "mpv_talloc.h"
#include "common/msg.h"
#include "common/global.h"
#include "osdep/threads.h"
#include "stream/stream.h"
#include "demux.h"
Rewrite ordered chapters and timeline stuff This uses a different method to piece segments together. The old approach basically changes to a new file (with a new start offset) any time a segment ends. This meant waiting for audio/video end on segment end, and then changing to the new segment all at once. It had a very weird impact on the playback core, and some things (like truly gapless segment transitions, or frame backstepping) just didn't work. The new approach adds the demux_timeline pseudo-demuxer, which presents an uniform packet stream from the many segments. This is pretty similar to how ordered chapters are implemented everywhere else. It also reminds of the FFmpeg concat pseudo-demuxer. The "pure" version of this approach doesn't work though. Segments can actually have different codec configurations (different extradata), and subtitles are most likely broken too. (Subtitles have multiple corner cases which break the pure stream-concatenation approach completely.) To counter this, we do two things: - Reinit the decoder with each segment. We go as far as allowing concatenating files with completely different codecs for the sake of EDL (which also uses the timeline infrastructure). A "lighter" approach would try to make use of decoder mechanism to update e.g. the extradata, but that seems fragile. - Clip decoded data to segment boundaries. This is equivalent to normal playback core mechanisms like hr-seek, but now the playback core doesn't need to care about these things. These two mechanisms are equivalent to what happened in the old implementation, except they don't happen in the playback core anymore. In other words, the playback core is completely relieved from timeline implementation details. (Which honestly is exactly what I'm trying to do here. I don't think ordered chapter behavior deserves improvement, even if it's bad - but I want to get it out from the playback core.) There is code duplication between audio and video decoder common code. This is awful and could be shareable - but this will happen later. Note that the audio path has some code to clip audio frames for the purpose of codec preroll/gapless handling, but it's not shared as sharing it would cause more pain than it would help.
2016-02-15 21:04:07 +01:00
#include "timeline.h"
#include "stheader.h"
#include "cue.h"
// Demuxer list
EDL: add support for new EDL file format The timeline code previously added to support Matroska ordered chapters allows constructing a playback timeline from segments picked from multiple source files. Add support for a new EDL format to make this machinery available for use with file formats other than Matroska and in a manner easier to use than creating files with ordered chapters. Unlike the old -edl option which specifies an additional file with edits to apply to the video file given as the main argument, the new EDL format is used by giving only the EDL file as the file to play; that file then contains the filename(s) to use as source files where actual video segments come from. Filename paths in the EDL file are ignored. Currently the source files are only searched for in the directory of the EDL file; support for a search path option will likely be added in the future. Format of the EDL files The first line in the file must be "mplayer EDL file, version 2". The rest of the lines belong to one of these classes: 1) lines specifying source files 2) empty lines 3) lines specifying timeline segments. Lines beginning with '<' specify source files. These lines first contain an identifier used to refer to the source file later, then the filename separated by whitespace. The identifier must start with a letter. Filenames that start or end with whitespace or contain newlines are not supported. On other lines '#' characters delimit comments. Lines that contain only whitespace after comments have been removed are ignored. Timeline segments must appear in the file in chronological order. Each segment has the following information associated with it: - duration - output start time - output end time (= output start time + duration) - source id (specifies the file the content of the segment comes from) - source start time (timestamp in the source file) - source end time (= source start time + duration) The output timestamps must form a continuous timeline from 0 to the end of the last segment, such that each new segment starts from the time the previous one ends at. Source files and times may change arbitrarily between segments. The general format for lines specifying timeline segments is [output time info] source_id [source time info] source_id must be an identifier defined on a '<' line. Both the time info parts consists of zero or more of the following elements: 1) timestamp 2) -timestamp 3) +duration 4) * 5) -* , where "timestamp" and "duration" are decimal numbers (computations are done with nanosecond precision). Whitespace around "+" and "-" is optional. 1) and 2) specify start and end time of the segment on output or source side. 3) specifies duration; the semantics are the same whether this appears on output or source side. 4) and 5) are ignored on the output side (they're always implicitly assumed). On the source side 4) specifies that the segment starts where the previous segment _using this source_ ended; if there was no previous segment time 0 is used. 5) specifies that the segment ends where the next segment using this source starts. Redundant information may be omitted. It will be filled in using the following rules: - output start for first segment is 0 - two of [output start, output end, duration] imply third - two of [source start, source end, duration] imply third - output start = output end of previous segment - output end = output start of next segment - if "*", source start = source end of earlier segment - if "-*", source end = source start of a later segment As a special rule, a last zero-duration segment without a source specification may appear. This will produce no corresponding segment in the resulting timeline, but can be used as syntax to specify the end time of the timeline (with effect equal to adding -time on the previous line). Examples: ----- begin ----- mplayer EDL file, version 2 < id1 filename 0 id1 123 100 id1 456 200 id1 789 300 ----- end ----- All segments come from the source file "filename". First segment (output time 0-100) comes from time 123-223, second 456-556, third 789-889. ----- begin ----- mplayer EDL file, version 2 < f filename f 60-120 f 600-660 f 30- 90 ----- end ----- Play first seconds 60-120 from the file, then 600-660, then 30-90. ----- begin ----- mplayer EDL file, version 2 < id1 filename1 < id2 filename2 +10 id1 * +10 id2 * +10 id1 * +10 id2 * +10 id1 * +10 id2 * ----- end ----- This plays time 0-10 from filename1, then 0-10 from filename1, then 10-20 from filename1, then 10-20 from filename2, then 20-30 from filename1, then 20-30 from filename2. ----- begin ----- mplayer EDL file, version 2 < t1 filename1 < t2 filename2 t1 * +2 # segment 1 +2 t2 100 # segment 2 t1 * # segment 3 t2 *-* # segment 4 t1 3 -* # segment 5 +0.111111 t2 102.5 # segment 6 7.37 t1 5 +1 # segment 7 ----- end ----- This rather pathological example illustrates the rules for filling in implied data. All the values can be determined by recursively applying the rules given above, and the full end result is this: +2 0-2 t1 0-2 # segment 1 +2 2-4 t2 100-102 # segment 2 +0.758889 4-4.758889 t1 2-2.758889 # segment 3 +0.5 4.4758889-5.258889 t2 102-102.5 # segment 4 +2 5.258889-7.258889 t1 3-5 # segment 5 +0.111111 7.258889-7.37 t2 102.5-102.611111 # segment 6 +1 7.37-8.37 t1 5-6 # segment 7
2011-02-14 12:05:35 +01:00
extern const struct demuxer_desc demuxer_desc_edl;
extern const struct demuxer_desc demuxer_desc_cue;
extern const demuxer_desc_t demuxer_desc_rawaudio;
extern const demuxer_desc_t demuxer_desc_rawvideo;
extern const demuxer_desc_t demuxer_desc_tv;
extern const demuxer_desc_t demuxer_desc_mf;
extern const demuxer_desc_t demuxer_desc_matroska;
extern const demuxer_desc_t demuxer_desc_lavf;
extern const demuxer_desc_t demuxer_desc_playlist;
extern const demuxer_desc_t demuxer_desc_disc;
extern const demuxer_desc_t demuxer_desc_rar;
extern const demuxer_desc_t demuxer_desc_libarchive;
extern const demuxer_desc_t demuxer_desc_null;
Rewrite ordered chapters and timeline stuff This uses a different method to piece segments together. The old approach basically changes to a new file (with a new start offset) any time a segment ends. This meant waiting for audio/video end on segment end, and then changing to the new segment all at once. It had a very weird impact on the playback core, and some things (like truly gapless segment transitions, or frame backstepping) just didn't work. The new approach adds the demux_timeline pseudo-demuxer, which presents an uniform packet stream from the many segments. This is pretty similar to how ordered chapters are implemented everywhere else. It also reminds of the FFmpeg concat pseudo-demuxer. The "pure" version of this approach doesn't work though. Segments can actually have different codec configurations (different extradata), and subtitles are most likely broken too. (Subtitles have multiple corner cases which break the pure stream-concatenation approach completely.) To counter this, we do two things: - Reinit the decoder with each segment. We go as far as allowing concatenating files with completely different codecs for the sake of EDL (which also uses the timeline infrastructure). A "lighter" approach would try to make use of decoder mechanism to update e.g. the extradata, but that seems fragile. - Clip decoded data to segment boundaries. This is equivalent to normal playback core mechanisms like hr-seek, but now the playback core doesn't need to care about these things. These two mechanisms are equivalent to what happened in the old implementation, except they don't happen in the playback core anymore. In other words, the playback core is completely relieved from timeline implementation details. (Which honestly is exactly what I'm trying to do here. I don't think ordered chapter behavior deserves improvement, even if it's bad - but I want to get it out from the playback core.) There is code duplication between audio and video decoder common code. This is awful and could be shareable - but this will happen later. Note that the audio path has some code to clip audio frames for the purpose of codec preroll/gapless handling, but it's not shared as sharing it would cause more pain than it would help.
2016-02-15 21:04:07 +01:00
extern const demuxer_desc_t demuxer_desc_timeline;
/* Please do not add any new demuxers here. If you want to implement a new
* demuxer, add it to libavformat, except for wrappers around external
* libraries and demuxers requiring binary support. */
const demuxer_desc_t *const demuxer_list[] = {
&demuxer_desc_disc,
EDL: add support for new EDL file format The timeline code previously added to support Matroska ordered chapters allows constructing a playback timeline from segments picked from multiple source files. Add support for a new EDL format to make this machinery available for use with file formats other than Matroska and in a manner easier to use than creating files with ordered chapters. Unlike the old -edl option which specifies an additional file with edits to apply to the video file given as the main argument, the new EDL format is used by giving only the EDL file as the file to play; that file then contains the filename(s) to use as source files where actual video segments come from. Filename paths in the EDL file are ignored. Currently the source files are only searched for in the directory of the EDL file; support for a search path option will likely be added in the future. Format of the EDL files The first line in the file must be "mplayer EDL file, version 2". The rest of the lines belong to one of these classes: 1) lines specifying source files 2) empty lines 3) lines specifying timeline segments. Lines beginning with '<' specify source files. These lines first contain an identifier used to refer to the source file later, then the filename separated by whitespace. The identifier must start with a letter. Filenames that start or end with whitespace or contain newlines are not supported. On other lines '#' characters delimit comments. Lines that contain only whitespace after comments have been removed are ignored. Timeline segments must appear in the file in chronological order. Each segment has the following information associated with it: - duration - output start time - output end time (= output start time + duration) - source id (specifies the file the content of the segment comes from) - source start time (timestamp in the source file) - source end time (= source start time + duration) The output timestamps must form a continuous timeline from 0 to the end of the last segment, such that each new segment starts from the time the previous one ends at. Source files and times may change arbitrarily between segments. The general format for lines specifying timeline segments is [output time info] source_id [source time info] source_id must be an identifier defined on a '<' line. Both the time info parts consists of zero or more of the following elements: 1) timestamp 2) -timestamp 3) +duration 4) * 5) -* , where "timestamp" and "duration" are decimal numbers (computations are done with nanosecond precision). Whitespace around "+" and "-" is optional. 1) and 2) specify start and end time of the segment on output or source side. 3) specifies duration; the semantics are the same whether this appears on output or source side. 4) and 5) are ignored on the output side (they're always implicitly assumed). On the source side 4) specifies that the segment starts where the previous segment _using this source_ ended; if there was no previous segment time 0 is used. 5) specifies that the segment ends where the next segment using this source starts. Redundant information may be omitted. It will be filled in using the following rules: - output start for first segment is 0 - two of [output start, output end, duration] imply third - two of [source start, source end, duration] imply third - output start = output end of previous segment - output end = output start of next segment - if "*", source start = source end of earlier segment - if "-*", source end = source start of a later segment As a special rule, a last zero-duration segment without a source specification may appear. This will produce no corresponding segment in the resulting timeline, but can be used as syntax to specify the end time of the timeline (with effect equal to adding -time on the previous line). Examples: ----- begin ----- mplayer EDL file, version 2 < id1 filename 0 id1 123 100 id1 456 200 id1 789 300 ----- end ----- All segments come from the source file "filename". First segment (output time 0-100) comes from time 123-223, second 456-556, third 789-889. ----- begin ----- mplayer EDL file, version 2 < f filename f 60-120 f 600-660 f 30- 90 ----- end ----- Play first seconds 60-120 from the file, then 600-660, then 30-90. ----- begin ----- mplayer EDL file, version 2 < id1 filename1 < id2 filename2 +10 id1 * +10 id2 * +10 id1 * +10 id2 * +10 id1 * +10 id2 * ----- end ----- This plays time 0-10 from filename1, then 0-10 from filename1, then 10-20 from filename1, then 10-20 from filename2, then 20-30 from filename1, then 20-30 from filename2. ----- begin ----- mplayer EDL file, version 2 < t1 filename1 < t2 filename2 t1 * +2 # segment 1 +2 t2 100 # segment 2 t1 * # segment 3 t2 *-* # segment 4 t1 3 -* # segment 5 +0.111111 t2 102.5 # segment 6 7.37 t1 5 +1 # segment 7 ----- end ----- This rather pathological example illustrates the rules for filling in implied data. All the values can be determined by recursively applying the rules given above, and the full end result is this: +2 0-2 t1 0-2 # segment 1 +2 2-4 t2 100-102 # segment 2 +0.758889 4-4.758889 t1 2-2.758889 # segment 3 +0.5 4.4758889-5.258889 t2 102-102.5 # segment 4 +2 5.258889-7.258889 t1 3-5 # segment 5 +0.111111 7.258889-7.37 t2 102.5-102.611111 # segment 6 +1 7.37-8.37 t1 5-6 # segment 7
2011-02-14 12:05:35 +01:00
&demuxer_desc_edl,
&demuxer_desc_cue,
&demuxer_desc_rawaudio,
&demuxer_desc_rawvideo,
#if HAVE_TV
&demuxer_desc_tv,
#endif
&demuxer_desc_matroska,
#if HAVE_LIBARCHIVE
&demuxer_desc_libarchive,
#endif
&demuxer_desc_rar,
&demuxer_desc_lavf,
&demuxer_desc_mf,
&demuxer_desc_playlist,
&demuxer_desc_null,
NULL
};
struct demux_internal {
struct mp_log *log;
// The demuxer runs potentially in another thread, so we keep two demuxer
// structs; the real demuxer can access the shadow struct only.
// Since demuxer and user threads both don't use locks, a third demuxer
// struct d_buffer is used to copy data between them in a synchronized way.
struct demuxer *d_thread; // accessed by demuxer impl. (producer)
struct demuxer *d_user; // accessed by player (consumer)
struct demuxer *d_buffer; // protected by lock; used to sync d_user/thread
// The lock protects the packet queues (struct demux_stream), d_buffer,
// and the fields below.
pthread_mutex_t lock;
pthread_cond_t wakeup;
pthread_t thread;
// -- All the following fields are protected by lock.
bool thread_terminate;
bool threading;
void (*wakeup_cb)(void *ctx);
void *wakeup_cb_ctx;
struct sh_stream **streams;
int num_streams;
int events;
bool warned_queue_overflow;
bool last_eof; // last actual global EOF status
bool eof; // whether we're in EOF state (reset for retry)
bool idle;
bool autoselect;
double min_secs;
int max_packs;
int max_bytes;
// Set if we know that we are at the start of the file. This is used to
// avoid a redundant initial seek after enabling streams. We could just
// allow it, but to avoid buggy seeking affecting normal playback, we don't.
bool initial_state;
bool tracks_switched; // thread needs to inform demuxer of this
bool seeking; // there's a seek queued
int seek_flags; // flags for next seek (if seeking==true)
double seek_pts;
double ref_pts; // assumed player position (only for track switches)
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
double ts_offset; // timestamp offset to apply to everything
void (*run_fn)(void *); // if non-NULL, function queued to be run on
void *run_fn_arg; // the thread as run_fn(run_fn_arg)
// Cached state.
bool force_cache_update;
double time_length;
struct mp_tags *stream_metadata;
struct stream_cache_info stream_cache_info;
int64_t stream_size;
// Updated during init only.
char *stream_base_filename;
};
2013-07-11 19:20:25 +02:00
struct demux_stream {
struct demux_internal *in;
enum stream_type type;
// all fields are protected by in->lock
bool selected; // user wants packets from this stream
bool active; // try to keep at least 1 packet queued
// if false, this stream is disabled, or passively
// read (like subtitles)
bool eof; // end of demuxed stream? (true if all buffer empty)
bool need_refresh; // enabled mid-stream
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
bool refreshing;
size_t packs; // number of packets in buffer
size_t bytes; // total bytes of packets in buffer
double base_ts; // timestamp of the last packet returned to decoder
double last_ts; // timestamp of the last packet added to queue
double last_br_ts; // timestamp of last packet bitrate was calculated
size_t last_br_bytes; // summed packet sizes since last bitrate calculation
double bitrate;
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
int64_t last_pos;
struct demux_packet *head;
struct demux_packet *tail;
// for closed captions (demuxer_feed_caption)
struct sh_stream *cc;
2013-07-11 19:20:25 +02:00
};
// Return "a", or if that is NOPTS, return "def".
#define PTS_OR_DEF(a, def) ((a) == MP_NOPTS_VALUE ? (def) : (a))
// If one of the values is NOPTS, always pick the other one.
#define MP_PTS_MIN(a, b) MPMIN(PTS_OR_DEF(a, b), PTS_OR_DEF(b, a))
#define MP_PTS_MAX(a, b) MPMAX(PTS_OR_DEF(a, b), PTS_OR_DEF(b, a))
#define MP_ADD_PTS(a, b) ((a) == MP_NOPTS_VALUE ? (a) : ((a) + (b)))
static void demuxer_sort_chapters(demuxer_t *demuxer);
static void *demux_thread(void *pctx);
static void update_cache(struct demux_internal *in);
// called locked
static void ds_flush(struct demux_stream *ds)
2013-07-11 19:20:25 +02:00
{
demux_packet_t *dp = ds->head;
while (dp) {
demux_packet_t *dn = dp->next;
free_demux_packet(dp);
dp = dn;
}
ds->head = ds->tail = NULL;
ds->packs = 0;
2013-07-11 19:20:25 +02:00
ds->bytes = 0;
ds->last_ts = ds->base_ts = ds->last_br_ts = MP_NOPTS_VALUE;
ds->last_br_bytes = 0;
ds->bitrate = -1;
ds->eof = false;
ds->active = false;
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
ds->refreshing = false;
ds->need_refresh = false;
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
ds->last_pos = -1;
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}
void demux_set_ts_offset(struct demuxer *demuxer, double offset)
{
struct demux_internal *in = demuxer->in;
pthread_mutex_lock(&in->lock);
in->ts_offset = offset;
pthread_mutex_unlock(&in->lock);
}
// Allocate a new sh_stream of the given type. It either has to be released
// with talloc_free(), or added to a demuxer with demux_add_sh_stream(). You
// cannot add or read packets from the stream before it has been added.
struct sh_stream *demux_alloc_sh_stream(enum stream_type type)
{
struct sh_stream *sh = talloc_ptrtype(NULL, sh);
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*sh = (struct sh_stream) {
.type = type,
.index = -1,
.ff_index = -1, // may be overwritten by demuxer
.demuxer_id = -1, // ... same
.codec = talloc_zero(sh, struct mp_codec_params),
};
sh->codec->type = type;
return sh;
}
// Add a new sh_stream to the demuxer. Note that as soon as the stream has been
// added, it must be immutable, and must not be released (this will happen when
// the demuxer is destroyed).
void demux_add_sh_stream(struct demuxer *demuxer, struct sh_stream *sh)
{
struct demux_internal *in = demuxer->in;
pthread_mutex_lock(&in->lock);
assert(!sh->ds); // must not be added yet
sh->ds = talloc(sh, struct demux_stream);
*sh->ds = (struct demux_stream) {
.in = in,
.type = sh->type,
.selected = in->autoselect,
2013-07-11 19:20:25 +02:00
};
if (!sh->codec->codec)
sh->codec->codec = "";
sh->index = in->num_streams;
if (sh->ff_index < 0)
sh->ff_index = sh->index;
if (sh->demuxer_id < 0) {
sh->demuxer_id = 0;
for (int n = 0; n < in->num_streams; n++) {
if (in->streams[n]->type == sh->type)
sh->demuxer_id += 1;
}
}
MP_TARRAY_APPEND(in, in->streams, in->num_streams, sh);
in->events |= DEMUX_EVENT_STREAMS;
if (in->wakeup_cb)
in->wakeup_cb(in->wakeup_cb_ctx);
pthread_mutex_unlock(&in->lock);
}
// Return a stream with the given index. Since streams can only be added during
// the lifetime of the demuxer, it is guaranteed that an index within the valid
// range [0, demux_get_num_stream()) always returns a valid sh_stream pointer,
// which will be valid until the demuxer is destroyed.
struct sh_stream *demux_get_stream(struct demuxer *demuxer, int index)
{
struct demux_internal *in = demuxer->in;
pthread_mutex_lock(&in->lock);
assert(index >= 0 && index < in->num_streams);
struct sh_stream *r = in->streams[index];
pthread_mutex_unlock(&in->lock);
return r;
}
// See demux_get_stream().
int demux_get_num_stream(struct demuxer *demuxer)
{
struct demux_internal *in = demuxer->in;
pthread_mutex_lock(&in->lock);
int r = in->num_streams;
pthread_mutex_unlock(&in->lock);
return r;
}
void free_demuxer(demuxer_t *demuxer)
{
if (!demuxer)
return;
struct demux_internal *in = demuxer->in;
assert(demuxer == in->d_user);
demux_stop_thread(demuxer);
if (demuxer->desc->close)
demuxer->desc->close(in->d_thread);
for (int n = in->num_streams - 1; n >= 0; n--) {
ds_flush(in->streams[n]->ds);
talloc_free(in->streams[n]);
}
pthread_mutex_destroy(&in->lock);
pthread_cond_destroy(&in->wakeup);
talloc_free(demuxer);
}
void free_demuxer_and_stream(struct demuxer *demuxer)
{
if (!demuxer)
return;
struct stream *s = demuxer->stream;
free_demuxer(demuxer);
free_stream(s);
}
// Start the demuxer thread, which reads ahead packets on its own.
void demux_start_thread(struct demuxer *demuxer)
{
struct demux_internal *in = demuxer->in;
assert(demuxer == in->d_user);
if (!in->threading) {
in->threading = true;
if (pthread_create(&in->thread, NULL, demux_thread, in))
in->threading = false;
}
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}
void demux_stop_thread(struct demuxer *demuxer)
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{
struct demux_internal *in = demuxer->in;
assert(demuxer == in->d_user);
if (in->threading) {
pthread_mutex_lock(&in->lock);
in->thread_terminate = true;
pthread_cond_signal(&in->wakeup);
pthread_mutex_unlock(&in->lock);
pthread_join(in->thread, NULL);
in->threading = false;
in->thread_terminate = false;
}
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}
// The demuxer thread will call cb(ctx) if there's a new packet, or EOF is reached.
void demux_set_wakeup_cb(struct demuxer *demuxer, void (*cb)(void *ctx), void *ctx)
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{
struct demux_internal *in = demuxer->in;
pthread_mutex_lock(&in->lock);
in->wakeup_cb = cb;
in->wakeup_cb_ctx = ctx;
pthread_mutex_unlock(&in->lock);
}
const char *stream_type_name(enum stream_type type)
{
switch (type) {
case STREAM_VIDEO: return "video";
case STREAM_AUDIO: return "audio";
case STREAM_SUB: return "sub";
default: return "unknown";
}
}
void demuxer_feed_caption(struct sh_stream *stream, demux_packet_t *dp)
{
struct demuxer *demuxer = stream->ds->in->d_thread;
struct sh_stream *sh = stream->ds->cc;
if (!sh) {
sh = demux_alloc_sh_stream(STREAM_SUB);
if (!sh) {
talloc_free(dp);
return;
}
sh->codec->codec = "eia_608";
stream->ds->cc = sh;
demux_add_sh_stream(demuxer, sh);
}
dp->pts = MP_ADD_PTS(dp->pts, -stream->ds->in->ts_offset);
dp->dts = MP_ADD_PTS(dp->dts, -stream->ds->in->ts_offset);
demux_add_packet(sh, dp);
}
// An obscure mechanism to get stream switching to be executed faster.
// On a switch, it seeks back, and then grabs all packets that were
// "missing" from the packet queue of the newly selected stream.
// Returns MP_NOPTS_VALUE if no seek should happen.
static double get_refresh_seek_pts(struct demux_internal *in)
{
struct demuxer *demux = in->d_thread;
double start_ts = in->ref_pts;
bool needed = false;
bool normal_seek = true;
for (int n = 0; n < in->num_streams; n++) {
struct demux_stream *ds = in->streams[n]->ds;
if (ds->type == STREAM_VIDEO || ds->type == STREAM_AUDIO)
start_ts = MP_PTS_MIN(start_ts, ds->base_ts);
needed |= ds->need_refresh;
// If there were no other streams selected, we can use a normal seek.
normal_seek &= ds->need_refresh || !ds->selected;
ds->need_refresh = false;
}
if (!needed || start_ts == MP_NOPTS_VALUE || !demux->desc->seek ||
!demux->seekable || demux->partially_seekable)
return MP_NOPTS_VALUE;
if (normal_seek)
return start_ts;
if (!demux->allow_refresh_seeks)
return MP_NOPTS_VALUE;
for (int n = 0; n < in->num_streams; n++) {
struct demux_stream *ds = in->streams[n]->ds;
// Streams which didn't read any packets yet can return all packets,
// or they'd be stuck forever; affects newly selected streams too.
if (ds->last_pos != -1)
ds->refreshing = true;
}
// Seek back to player's current position, with a small offset added.
return start_ts - 1.0;
}
void demux_add_packet(struct sh_stream *stream, demux_packet_t *dp)
{
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struct demux_stream *ds = stream ? stream->ds : NULL;
if (!dp || !ds) {
talloc_free(dp);
return;
}
struct demux_internal *in = ds->in;
pthread_mutex_lock(&in->lock);
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
bool drop = false;
if (ds->refreshing) {
// Resume reading once the old position was reached (i.e. we start
// returning packets where we left off before the refresh).
drop = dp->pos <= ds->last_pos;
if (dp->pos >= ds->last_pos)
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
ds->refreshing = false;
}
if (!ds->selected || in->seeking || drop) {
pthread_mutex_unlock(&in->lock);
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talloc_free(dp);
return;
2013-07-11 19:20:25 +02:00
}
dp->stream = stream->index;
dp->next = NULL;
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
ds->last_pos = dp->pos;
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ds->packs++;
ds->bytes += dp->len;
if (ds->tail) {
// next packet in stream
ds->tail->next = dp;
ds->tail = dp;
} else {
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// first packet in stream
ds->head = ds->tail = dp;
}
// obviously not true anymore
ds->eof = false;
in->last_eof = in->eof = false;
// For video, PTS determination is not trivial, but for other media types
// distinguishing PTS and DTS is not useful.
if (stream->type != STREAM_VIDEO && dp->pts == MP_NOPTS_VALUE)
dp->pts = dp->dts;
double ts = dp->dts == MP_NOPTS_VALUE ? dp->pts : dp->dts;
if (ts != MP_NOPTS_VALUE && (ts > ds->last_ts || ts + 10 < ds->last_ts))
ds->last_ts = ts;
if (ds->base_ts == MP_NOPTS_VALUE)
ds->base_ts = ds->last_ts;
MP_DBG(in, "append packet to %s: size=%d pts=%f dts=%f pos=%"PRIi64" "
"[num=%zd size=%zd]\n", stream_type_name(stream->type),
dp->len, dp->pts, dp->dts, dp->pos, ds->packs, ds->bytes);
if (ds->in->wakeup_cb && !ds->head->next)
ds->in->wakeup_cb(ds->in->wakeup_cb_ctx);
pthread_cond_signal(&in->wakeup);
pthread_mutex_unlock(&in->lock);
}
// Returns true if there was "progress" (lock was released temporarily).
static bool read_packet(struct demux_internal *in)
{
in->eof = false;
in->idle = true;
// Check if we need to read a new packet. We do this if all queues are below
// the minimum, or if a stream explicitly needs new packets. Also includes
// safe-guards against packet queue overflow.
bool active = false, read_more = false;
size_t packs = 0, bytes = 0;
for (int n = 0; n < in->num_streams; n++) {
struct demux_stream *ds = in->streams[n]->ds;
active |= ds->active;
read_more |= ds->active && !ds->head;
packs += ds->packs;
bytes += ds->bytes;
if (ds->active && ds->last_ts != MP_NOPTS_VALUE && in->min_secs > 0 &&
ds->last_ts >= ds->base_ts)
read_more |= ds->last_ts - ds->base_ts < in->min_secs;
2013-07-11 19:20:25 +02:00
}
MP_DBG(in, "packets=%zd, bytes=%zd, active=%d, more=%d\n",
packs, bytes, active, read_more);
if (packs >= in->max_packs || bytes >= in->max_bytes) {
if (!in->warned_queue_overflow) {
in->warned_queue_overflow = true;
MP_ERR(in, "Too many packets in the demuxer packet queues:\n");
for (int n = 0; n < in->num_streams; n++) {
struct demux_stream *ds = in->streams[n]->ds;
if (ds->selected) {
MP_ERR(in, " %s/%d: %zd packets, %zd bytes\n",
stream_type_name(ds->type), n, ds->packs, ds->bytes);
}
}
}
for (int n = 0; n < in->num_streams; n++) {
struct demux_stream *ds = in->streams[n]->ds;
bool eof = !ds->head;
if (eof && !ds->eof) {
if (in->wakeup_cb)
in->wakeup_cb(in->wakeup_cb_ctx);
}
ds->eof |= eof;
}
pthread_cond_signal(&in->wakeup);
return false;
}
if (!read_more)
return false;
double seek_pts = get_refresh_seek_pts(in);
// Actually read a packet. Drop the lock while doing so, because waiting
// for disk or network I/O can take time.
in->idle = false;
in->initial_state = false;
pthread_mutex_unlock(&in->lock);
struct demuxer *demux = in->d_thread;
if (seek_pts != MP_NOPTS_VALUE) {
MP_VERBOSE(in, "refresh seek to %f\n", seek_pts);
demux->desc->seek(demux, seek_pts, SEEK_BACKWARD | SEEK_HR);
}
bool eof = !demux->desc->fill_buffer || demux->desc->fill_buffer(demux) <= 0;
update_cache(in);
pthread_mutex_lock(&in->lock);
if (eof) {
for (int n = 0; n < in->num_streams; n++)
in->streams[n]->ds->eof = true;
// If we had EOF previously, then don't wakeup (avoids wakeup loop)
if (!in->last_eof) {
if (in->wakeup_cb)
in->wakeup_cb(in->wakeup_cb_ctx);
pthread_cond_signal(&in->wakeup);
MP_VERBOSE(in, "EOF reached.\n");
}
}
in->eof = in->last_eof = eof;
return true;
}
// must be called locked; may temporarily unlock
static void ds_get_packets(struct demux_stream *ds)
{
const char *t = stream_type_name(ds->type);
struct demux_internal *in = ds->in;
MP_DBG(in, "reading packet for %s\n", t);
in->eof = false; // force retry
ds->eof = false;
while (ds->selected && !ds->head && !ds->eof) {
ds->active = true;
// Note: the following code marks EOF if it can't continue
if (in->threading) {
MP_VERBOSE(in, "waiting for demux thread (%s)\n", t);
pthread_cond_signal(&in->wakeup);
pthread_cond_wait(&in->wakeup, &in->lock);
} else {
read_packet(in);
}
}
}
static void execute_trackswitch(struct demux_internal *in)
{
in->tracks_switched = false;
bool any_selected = false;
for (int n = 0; n < in->num_streams; n++)
any_selected |= in->streams[n]->ds->selected;
pthread_mutex_unlock(&in->lock);
if (in->d_thread->desc->control)
in->d_thread->desc->control(in->d_thread, DEMUXER_CTRL_SWITCHED_TRACKS, 0);
stream_control(in->d_thread->stream, STREAM_CTRL_SET_READAHEAD,
&(int){any_selected});
pthread_mutex_lock(&in->lock);
}
static void execute_seek(struct demux_internal *in)
{
int flags = in->seek_flags;
double pts = in->seek_pts;
in->seeking = false;
in->initial_state = false;
pthread_mutex_unlock(&in->lock);
MP_VERBOSE(in, "execute seek (to %f flags %d)\n", pts, flags);
if (in->d_thread->desc->seek)
in->d_thread->desc->seek(in->d_thread, pts, flags);
MP_VERBOSE(in, "seek done\n");
pthread_mutex_lock(&in->lock);
}
static void *demux_thread(void *pctx)
{
struct demux_internal *in = pctx;
mpthread_set_name("demux");
pthread_mutex_lock(&in->lock);
while (!in->thread_terminate) {
if (in->run_fn) {
in->run_fn(in->run_fn_arg);
in->run_fn = NULL;
pthread_cond_signal(&in->wakeup);
continue;
}
if (in->tracks_switched) {
execute_trackswitch(in);
continue;
}
if (in->seeking) {
execute_seek(in);
continue;
}
if (!in->eof) {
if (read_packet(in))
continue; // read_packet unlocked, so recheck conditions
}
if (in->force_cache_update) {
pthread_mutex_unlock(&in->lock);
update_cache(in);
pthread_mutex_lock(&in->lock);
in->force_cache_update = false;
continue;
}
pthread_cond_signal(&in->wakeup);
pthread_cond_wait(&in->wakeup, &in->lock);
}
pthread_mutex_unlock(&in->lock);
return NULL;
}
static struct demux_packet *dequeue_packet(struct demux_stream *ds)
{
if (!ds->head)
return NULL;
struct demux_packet *pkt = ds->head;
ds->head = pkt->next;
pkt->next = NULL;
if (!ds->head)
ds->tail = NULL;
ds->bytes -= pkt->len;
ds->packs--;
double ts = pkt->dts == MP_NOPTS_VALUE ? pkt->pts : pkt->dts;
if (ts != MP_NOPTS_VALUE)
ds->base_ts = ts;
if (pkt->keyframe && ts != MP_NOPTS_VALUE) {
// Update bitrate - only at keyframe points, because we use the
// (possibly) reordered packet timestamps instead of realtime.
double d = ts - ds->last_br_ts;
if (ds->last_br_ts == MP_NOPTS_VALUE || d < 0) {
ds->bitrate = -1;
ds->last_br_ts = ts;
ds->last_br_bytes = 0;
} else if (d > 0 && d >= 0.5) { // a window of least 500ms for UI purposes
ds->bitrate = ds->last_br_bytes / d;
ds->last_br_ts = ts;
ds->last_br_bytes = 0;
}
}
ds->last_br_bytes += pkt->len;
// This implies this function is actually called from "the" user thread.
if (pkt->pos >= ds->in->d_user->filepos)
ds->in->d_user->filepos = pkt->pos;
pkt->pts = MP_ADD_PTS(pkt->pts, ds->in->ts_offset);
pkt->dts = MP_ADD_PTS(pkt->dts, ds->in->ts_offset);
pkt->start = MP_ADD_PTS(pkt->start, ds->in->ts_offset);
pkt->end = MP_ADD_PTS(pkt->end, ds->in->ts_offset);
return pkt;
}
// Sparse packets (Subtitles) interleaved with other non-sparse packets (video,
// audio) should never be read actively, meaning the demuxer thread does not
// try to exceed default readahead in order to find a new packet.
static bool use_lazy_subtitle_reading(struct demux_stream *ds)
{
if (ds->type != STREAM_SUB)
return false;
for (int n = 0; n < ds->in->num_streams; n++) {
struct demux_stream *s = ds->in->streams[n]->ds;
if (s->type != STREAM_SUB && s->selected && !s->eof)
return true;
}
return false;
}
// Read a packet from the given stream. The returned packet belongs to the
// caller, who has to free it with talloc_free(). Might block. Returns NULL
// on EOF.
struct demux_packet *demux_read_packet(struct sh_stream *sh)
{
2013-07-11 19:20:25 +02:00
struct demux_stream *ds = sh ? sh->ds : NULL;
struct demux_packet *pkt = NULL;
if (ds) {
pthread_mutex_lock(&ds->in->lock);
if (!use_lazy_subtitle_reading(ds))
ds_get_packets(ds);
pkt = dequeue_packet(ds);
pthread_cond_signal(&ds->in->wakeup); // possibly read more
pthread_mutex_unlock(&ds->in->lock);
}
return pkt;
}
// Poll the demuxer queue, and if there's a packet, return it. Otherwise, just
// make the demuxer thread read packets for this stream, and if there's at
// least one packet, call the wakeup callback.
// Unlike demux_read_packet(), this always enables readahead (except for
// interleaved subtitles).
// Returns:
// < 0: EOF was reached, *out_pkt=NULL
// == 0: no new packet yet, but maybe later, *out_pkt=NULL
// > 0: new packet read, *out_pkt is set
// Note: when reading interleaved subtitles, the demuxer won't try to forcibly
// read ahead to get the next subtitle packet (as the next packet could be
// minutes away). In this situation, this function will just return -1.
int demux_read_packet_async(struct sh_stream *sh, struct demux_packet **out_pkt)
{
struct demux_stream *ds = sh ? sh->ds : NULL;
int r = -1;
*out_pkt = NULL;
if (ds) {
if (ds->in->threading) {
pthread_mutex_lock(&ds->in->lock);
*out_pkt = dequeue_packet(ds);
if (use_lazy_subtitle_reading(ds)) {
r = *out_pkt ? 1 : -1;
} else {
r = *out_pkt ? 1 : ((ds->eof || !ds->selected) ? -1 : 0);
ds->active = ds->selected; // enable readahead
ds->in->eof = false; // force retry
pthread_cond_signal(&ds->in->wakeup); // possibly read more
}
pthread_mutex_unlock(&ds->in->lock);
} else {
*out_pkt = demux_read_packet(sh);
r = *out_pkt ? 1 : -1;
}
}
return r;
}
// Return whether a packet is queued. Never blocks, never forces any reads.
bool demux_has_packet(struct sh_stream *sh)
{
bool has_packet = false;
if (sh) {
pthread_mutex_lock(&sh->ds->in->lock);
has_packet = sh->ds->head;
pthread_mutex_unlock(&sh->ds->in->lock);
}
return has_packet;
}
Rewrite ordered chapters and timeline stuff This uses a different method to piece segments together. The old approach basically changes to a new file (with a new start offset) any time a segment ends. This meant waiting for audio/video end on segment end, and then changing to the new segment all at once. It had a very weird impact on the playback core, and some things (like truly gapless segment transitions, or frame backstepping) just didn't work. The new approach adds the demux_timeline pseudo-demuxer, which presents an uniform packet stream from the many segments. This is pretty similar to how ordered chapters are implemented everywhere else. It also reminds of the FFmpeg concat pseudo-demuxer. The "pure" version of this approach doesn't work though. Segments can actually have different codec configurations (different extradata), and subtitles are most likely broken too. (Subtitles have multiple corner cases which break the pure stream-concatenation approach completely.) To counter this, we do two things: - Reinit the decoder with each segment. We go as far as allowing concatenating files with completely different codecs for the sake of EDL (which also uses the timeline infrastructure). A "lighter" approach would try to make use of decoder mechanism to update e.g. the extradata, but that seems fragile. - Clip decoded data to segment boundaries. This is equivalent to normal playback core mechanisms like hr-seek, but now the playback core doesn't need to care about these things. These two mechanisms are equivalent to what happened in the old implementation, except they don't happen in the playback core anymore. In other words, the playback core is completely relieved from timeline implementation details. (Which honestly is exactly what I'm trying to do here. I don't think ordered chapter behavior deserves improvement, even if it's bad - but I want to get it out from the playback core.) There is code duplication between audio and video decoder common code. This is awful and could be shareable - but this will happen later. Note that the audio path has some code to clip audio frames for the purpose of codec preroll/gapless handling, but it's not shared as sharing it would cause more pain than it would help.
2016-02-15 21:04:07 +01:00
// Read and return any packet we find. NULL means EOF.
struct demux_packet *demux_read_any_packet(struct demuxer *demuxer)
{
struct demux_internal *in = demuxer->in;
assert(!in->threading); // doesn't work with threading
bool read_more = true;
while (read_more) {
for (int n = 0; n < in->num_streams; n++) {
struct sh_stream *sh = in->streams[n];
sh->ds->active = sh->ds->selected; // force read_packet() to read
struct demux_packet *pkt = dequeue_packet(sh->ds);
if (pkt)
return pkt;
}
// retry after calling this
pthread_mutex_lock(&in->lock); // lock only because read_packet unlocks
read_more = read_packet(in);
read_more &= !in->eof;
pthread_mutex_unlock(&in->lock);
}
return NULL;
}
// ====================================================================
void demuxer_help(struct mp_log *log)
{
int i;
mp_info(log, "Available demuxers:\n");
mp_info(log, " demuxer: info:\n");
for (i = 0; demuxer_list[i]; i++) {
mp_info(log, "%10s %s\n",
demuxer_list[i]->name, demuxer_list[i]->desc);
}
}
static const char *d_level(enum demux_check level)
{
switch (level) {
case DEMUX_CHECK_FORCE: return "force";
case DEMUX_CHECK_UNSAFE: return "unsafe";
case DEMUX_CHECK_REQUEST:return "request";
case DEMUX_CHECK_NORMAL: return "normal";
}
abort();
}
static int decode_float(char *str, float *out)
{
char *rest;
float dec_val;
dec_val = strtod(str, &rest);
if (!rest || (rest == str) || !isfinite(dec_val))
return -1;
*out = dec_val;
return 0;
}
static int decode_gain(demuxer_t *demuxer, const char *tag, float *out)
{
char *tag_val = NULL;
float dec_val;
tag_val = mp_tags_get_str(demuxer->metadata, tag);
if (!tag_val)
return -1;
if (decode_float(tag_val, &dec_val)) {
mp_msg(demuxer->log, MSGL_ERR, "Invalid replaygain value\n");
return -1;
}
*out = dec_val;
return 0;
}
static int decode_peak(demuxer_t *demuxer, const char *tag, float *out)
{
char *tag_val = NULL;
float dec_val;
*out = 1.0;
tag_val = mp_tags_get_str(demuxer->metadata, tag);
if (!tag_val)
return 0;
if (decode_float(tag_val, &dec_val))
return 0;
if (dec_val == 0.0)
return 0;
*out = dec_val;
return 0;
}
static void apply_replaygain(demuxer_t *demuxer, struct replaygain_data *rg)
{
struct demux_internal *in = demuxer->in;
for (int n = 0; n < in->num_streams; n++) {
struct sh_stream *sh = in->streams[n];
if (sh->type == STREAM_AUDIO && !sh->codec->replaygain_data) {
MP_VERBOSE(demuxer, "Replaygain: Track=%f/%f Album=%f/%f\n",
rg->track_gain, rg->track_peak,
rg->album_gain, rg->album_peak);
sh->codec->replaygain_data = talloc_memdup(in, rg, sizeof(*rg));
}
}
}
static void demux_export_replaygain(demuxer_t *demuxer)
{
struct replaygain_data rg = {0};
if (!decode_gain(demuxer, "REPLAYGAIN_TRACK_GAIN", &rg.track_gain) &&
!decode_peak(demuxer, "REPLAYGAIN_TRACK_PEAK", &rg.track_peak) &&
!decode_gain(demuxer, "REPLAYGAIN_ALBUM_GAIN", &rg.album_gain) &&
!decode_peak(demuxer, "REPLAYGAIN_ALBUM_PEAK", &rg.album_peak))
{
apply_replaygain(demuxer, &rg);
}
if (!decode_gain(demuxer, "REPLAYGAIN_GAIN", &rg.track_gain) &&
!decode_peak(demuxer, "REPLAYGAIN_PEAK", &rg.track_peak))
{
rg.album_gain = rg.track_gain;
rg.album_peak = rg.track_peak;
apply_replaygain(demuxer, &rg);
}
}
// Copy all fields from src to dst, depending on event flags.
static void demux_copy(struct demuxer *dst, struct demuxer *src)
{
if (src->events & DEMUX_EVENT_INIT) {
// Note that we do as shallow copies as possible. We expect the data
// that is not-copied (only referenced) to be immutable.
// This implies e.g. that no chapters are added after initialization.
dst->chapters = src->chapters;
dst->num_chapters = src->num_chapters;
dst->editions = src->editions;
dst->num_editions = src->num_editions;
dst->edition = src->edition;
dst->attachments = src->attachments;
dst->num_attachments = src->num_attachments;
dst->matroska_data = src->matroska_data;
dst->playlist = src->playlist;
dst->seekable = src->seekable;
dst->partially_seekable = src->partially_seekable;
dst->filetype = src->filetype;
dst->ts_resets_possible = src->ts_resets_possible;
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
dst->allow_refresh_seeks = src->allow_refresh_seeks;
dst->fully_read = src->fully_read;
dst->start_time = src->start_time;
dst->priv = src->priv;
}
if (src->events & DEMUX_EVENT_METADATA) {
talloc_free(dst->metadata);
dst->metadata = mp_tags_dup(dst, src->metadata);
}
dst->events |= src->events;
src->events = 0;
}
// This is called by demuxer implementations if certain parameters change
// at runtime.
// events is one of DEMUX_EVENT_*
// The code will copy the fields references by the events to the user-thread.
void demux_changed(demuxer_t *demuxer, int events)
{
assert(demuxer == demuxer->in->d_thread); // call from demuxer impl. only
struct demux_internal *in = demuxer->in;
demuxer->events |= events;
update_cache(in);
pthread_mutex_lock(&in->lock);
if (demuxer->events & DEMUX_EVENT_INIT)
demuxer_sort_chapters(demuxer);
if (demuxer->events & (DEMUX_EVENT_METADATA | DEMUX_EVENT_STREAMS))
demux_export_replaygain(demuxer);
demux_copy(in->d_buffer, demuxer);
if (in->wakeup_cb)
in->wakeup_cb(in->wakeup_cb_ctx);
pthread_mutex_unlock(&in->lock);
}
// Called by the user thread (i.e. player) to update metadata and other things
// from the demuxer thread.
void demux_update(demuxer_t *demuxer)
{
assert(demuxer == demuxer->in->d_user);
struct demux_internal *in = demuxer->in;
if (!in->threading)
update_cache(in);
pthread_mutex_lock(&in->lock);
demux_copy(demuxer, in->d_buffer);
demuxer->events |= in->events;
in->events = 0;
if (in->stream_metadata && (demuxer->events & DEMUX_EVENT_METADATA))
mp_tags_merge(demuxer->metadata, in->stream_metadata);
pthread_mutex_unlock(&in->lock);
}
static void demux_init_cache(struct demuxer *demuxer)
{
struct demux_internal *in = demuxer->in;
struct stream *stream = demuxer->stream;
char *base = NULL;
stream_control(stream, STREAM_CTRL_GET_BASE_FILENAME, &base);
in->stream_base_filename = talloc_steal(demuxer, base);
}
static void demux_init_cuesheet(struct demuxer *demuxer)
{
char *cue = mp_tags_get_str(demuxer->metadata, "cuesheet");
if (cue && !demuxer->num_chapters) {
struct cue_file *f = mp_parse_cue(bstr0(cue));
if (f) {
if (mp_check_embedded_cue(f) < 0) {
MP_WARN(demuxer, "Embedded cue sheet references more than one file. "
"Ignoring it.\n");
} else {
for (int n = 0; n < f->num_tracks; n++) {
struct cue_track *t = &f->tracks[n];
int idx = demuxer_add_chapter(demuxer, "", t->start, -1);
mp_tags_merge(demuxer->chapters[idx].metadata, t->tags);
}
}
}
talloc_free(f);
}
}
static struct demuxer *open_given_type(struct mpv_global *global,
struct mp_log *log,
const struct demuxer_desc *desc,
struct stream *stream,
struct demuxer_params *params,
enum demux_check check)
{
if (mp_cancel_test(stream->cancel))
return NULL;
struct demuxer *demuxer = talloc_ptrtype(NULL, demuxer);
*demuxer = (struct demuxer) {
.desc = desc,
.stream = stream,
.seekable = stream->seekable,
.filepos = -1,
.opts = global->opts,
.global = global,
.log = mp_log_new(demuxer, log, desc->name),
.glog = log,
.filename = talloc_strdup(demuxer, stream->url),
.events = DEMUX_EVENT_ALL,
};
demuxer->seekable = stream->seekable;
if (demuxer->stream->uncached_stream &&
!demuxer->stream->uncached_stream->seekable)
demuxer->seekable = false;
struct demux_internal *in = demuxer->in = talloc_ptrtype(demuxer, in);
*in = (struct demux_internal){
.log = demuxer->log,
.d_thread = talloc(demuxer, struct demuxer),
.d_buffer = talloc(demuxer, struct demuxer),
.d_user = demuxer,
.min_secs = demuxer->opts->demuxer_min_secs,
.max_packs = demuxer->opts->demuxer_max_packs,
.max_bytes = demuxer->opts->demuxer_max_bytes,
.initial_state = true,
};
pthread_mutex_init(&in->lock, NULL);
pthread_cond_init(&in->wakeup, NULL);
if (stream->uncached_stream)
in->min_secs = MPMAX(in->min_secs, demuxer->opts->demuxer_min_secs_cache);
*in->d_thread = *demuxer;
*in->d_buffer = *demuxer;
in->d_thread->metadata = talloc_zero(in->d_thread, struct mp_tags);
in->d_user->metadata = talloc_zero(in->d_user, struct mp_tags);
in->d_buffer->metadata = talloc_zero(in->d_buffer, struct mp_tags);
mp_dbg(log, "Trying demuxer: %s (force-level: %s)\n",
desc->name, d_level(check));
Rewrite ordered chapters and timeline stuff This uses a different method to piece segments together. The old approach basically changes to a new file (with a new start offset) any time a segment ends. This meant waiting for audio/video end on segment end, and then changing to the new segment all at once. It had a very weird impact on the playback core, and some things (like truly gapless segment transitions, or frame backstepping) just didn't work. The new approach adds the demux_timeline pseudo-demuxer, which presents an uniform packet stream from the many segments. This is pretty similar to how ordered chapters are implemented everywhere else. It also reminds of the FFmpeg concat pseudo-demuxer. The "pure" version of this approach doesn't work though. Segments can actually have different codec configurations (different extradata), and subtitles are most likely broken too. (Subtitles have multiple corner cases which break the pure stream-concatenation approach completely.) To counter this, we do two things: - Reinit the decoder with each segment. We go as far as allowing concatenating files with completely different codecs for the sake of EDL (which also uses the timeline infrastructure). A "lighter" approach would try to make use of decoder mechanism to update e.g. the extradata, but that seems fragile. - Clip decoded data to segment boundaries. This is equivalent to normal playback core mechanisms like hr-seek, but now the playback core doesn't need to care about these things. These two mechanisms are equivalent to what happened in the old implementation, except they don't happen in the playback core anymore. In other words, the playback core is completely relieved from timeline implementation details. (Which honestly is exactly what I'm trying to do here. I don't think ordered chapter behavior deserves improvement, even if it's bad - but I want to get it out from the playback core.) There is code duplication between audio and video decoder common code. This is awful and could be shareable - but this will happen later. Note that the audio path has some code to clip audio frames for the purpose of codec preroll/gapless handling, but it's not shared as sharing it would cause more pain than it would help.
2016-02-15 21:04:07 +01:00
// not for DVD/BD/DVB in particular
if (stream->seekable && (!params || !params->timeline))
stream_seek(stream, 0);
// Peek this much data to avoid that stream_read() run by some demuxers
// will flush previous peeked data.
stream_peek(stream, STREAM_BUFFER_SIZE);
in->d_thread->params = params; // temporary during open()
int ret = demuxer->desc->open(in->d_thread, check);
if (ret >= 0) {
in->d_thread->params = NULL;
if (in->d_thread->filetype)
mp_verbose(log, "Detected file format: %s (%s)\n",
in->d_thread->filetype, desc->desc);
else
mp_verbose(log, "Detected file format: %s\n", desc->desc);
if (!in->d_thread->seekable)
mp_verbose(log, "Stream is not seekable.\n");
if (!in->d_thread->seekable && demuxer->opts->force_seekable) {
mp_warn(log, "Not seekable, but enabling seeking on user request.\n");
in->d_thread->seekable = true;
in->d_thread->partially_seekable = true;
}
demux_init_cuesheet(in->d_thread);
demux_init_cache(demuxer);
demux_changed(in->d_thread, DEMUX_EVENT_ALL);
demux_update(demuxer);
stream_control(demuxer->stream, STREAM_CTRL_SET_READAHEAD, &(int){false});
Rewrite ordered chapters and timeline stuff This uses a different method to piece segments together. The old approach basically changes to a new file (with a new start offset) any time a segment ends. This meant waiting for audio/video end on segment end, and then changing to the new segment all at once. It had a very weird impact on the playback core, and some things (like truly gapless segment transitions, or frame backstepping) just didn't work. The new approach adds the demux_timeline pseudo-demuxer, which presents an uniform packet stream from the many segments. This is pretty similar to how ordered chapters are implemented everywhere else. It also reminds of the FFmpeg concat pseudo-demuxer. The "pure" version of this approach doesn't work though. Segments can actually have different codec configurations (different extradata), and subtitles are most likely broken too. (Subtitles have multiple corner cases which break the pure stream-concatenation approach completely.) To counter this, we do two things: - Reinit the decoder with each segment. We go as far as allowing concatenating files with completely different codecs for the sake of EDL (which also uses the timeline infrastructure). A "lighter" approach would try to make use of decoder mechanism to update e.g. the extradata, but that seems fragile. - Clip decoded data to segment boundaries. This is equivalent to normal playback core mechanisms like hr-seek, but now the playback core doesn't need to care about these things. These two mechanisms are equivalent to what happened in the old implementation, except they don't happen in the playback core anymore. In other words, the playback core is completely relieved from timeline implementation details. (Which honestly is exactly what I'm trying to do here. I don't think ordered chapter behavior deserves improvement, even if it's bad - but I want to get it out from the playback core.) There is code duplication between audio and video decoder common code. This is awful and could be shareable - but this will happen later. Note that the audio path has some code to clip audio frames for the purpose of codec preroll/gapless handling, but it's not shared as sharing it would cause more pain than it would help.
2016-02-15 21:04:07 +01:00
struct timeline *tl = timeline_load(global, log, demuxer);
if (tl) {
struct demuxer_params params2 = {0};
params2.timeline = tl;
struct demuxer *sub = open_given_type(global, log,
&demuxer_desc_timeline, stream,
&params2, DEMUX_CHECK_FORCE);
if (sub)
return sub;
timeline_destroy(tl);
Rewrite ordered chapters and timeline stuff This uses a different method to piece segments together. The old approach basically changes to a new file (with a new start offset) any time a segment ends. This meant waiting for audio/video end on segment end, and then changing to the new segment all at once. It had a very weird impact on the playback core, and some things (like truly gapless segment transitions, or frame backstepping) just didn't work. The new approach adds the demux_timeline pseudo-demuxer, which presents an uniform packet stream from the many segments. This is pretty similar to how ordered chapters are implemented everywhere else. It also reminds of the FFmpeg concat pseudo-demuxer. The "pure" version of this approach doesn't work though. Segments can actually have different codec configurations (different extradata), and subtitles are most likely broken too. (Subtitles have multiple corner cases which break the pure stream-concatenation approach completely.) To counter this, we do two things: - Reinit the decoder with each segment. We go as far as allowing concatenating files with completely different codecs for the sake of EDL (which also uses the timeline infrastructure). A "lighter" approach would try to make use of decoder mechanism to update e.g. the extradata, but that seems fragile. - Clip decoded data to segment boundaries. This is equivalent to normal playback core mechanisms like hr-seek, but now the playback core doesn't need to care about these things. These two mechanisms are equivalent to what happened in the old implementation, except they don't happen in the playback core anymore. In other words, the playback core is completely relieved from timeline implementation details. (Which honestly is exactly what I'm trying to do here. I don't think ordered chapter behavior deserves improvement, even if it's bad - but I want to get it out from the playback core.) There is code duplication between audio and video decoder common code. This is awful and could be shareable - but this will happen later. Note that the audio path has some code to clip audio frames for the purpose of codec preroll/gapless handling, but it's not shared as sharing it would cause more pain than it would help.
2016-02-15 21:04:07 +01:00
}
return demuxer;
}
free_demuxer(demuxer);
return NULL;
}
static const int d_normal[] = {DEMUX_CHECK_NORMAL, DEMUX_CHECK_UNSAFE, -1};
static const int d_request[] = {DEMUX_CHECK_REQUEST, -1};
static const int d_force[] = {DEMUX_CHECK_FORCE, -1};
// params can be NULL
struct demuxer *demux_open(struct stream *stream, struct demuxer_params *params,
struct mpv_global *global)
{
const int *check_levels = d_normal;
const struct demuxer_desc *check_desc = NULL;
struct mp_log *log = mp_log_new(NULL, global->log, "!demux");
struct demuxer *demuxer = NULL;
char *force_format = params ? params->force_format : NULL;
if (!force_format)
force_format = stream->demuxer;
if (force_format && force_format[0]) {
check_levels = d_request;
if (force_format[0] == '+') {
force_format += 1;
check_levels = d_force;
}
for (int n = 0; demuxer_list[n]; n++) {
if (strcmp(demuxer_list[n]->name, force_format) == 0)
check_desc = demuxer_list[n];
}
if (!check_desc) {
mp_err(log, "Demuxer %s does not exist.\n", force_format);
goto done;
}
}
// Test demuxers from first to last, one pass for each check_levels[] entry
for (int pass = 0; check_levels[pass] != -1; pass++) {
enum demux_check level = check_levels[pass];
mp_verbose(log, "Trying demuxers for level=%s.\n", d_level(level));
for (int n = 0; demuxer_list[n]; n++) {
const struct demuxer_desc *desc = demuxer_list[n];
if (!check_desc || desc == check_desc) {
demuxer = open_given_type(global, log, desc, stream, params, level);
if (demuxer) {
talloc_steal(demuxer, log);
log = NULL;
goto done;
}
}
}
}
done:
talloc_free(log);
return demuxer;
}
// Convenience function: open the stream, enable the cache (according to params
// and global opts.), open the demuxer.
// (use free_demuxer_and_stream() to free the underlying stream too)
struct demuxer *demux_open_url(const char *url,
struct demuxer_params *params,
struct mp_cancel *cancel,
struct mpv_global *global)
{
struct MPOpts *opts = global->opts;
struct demuxer_params dummy = {0};
if (!params)
params = &dummy;
struct stream *s = stream_create(url, STREAM_READ | params->stream_flags,
cancel, global);
if (!s)
return NULL;
if (params->allow_capture)
stream_set_capture_file(s, opts->stream_capture);
if (!params->disable_cache)
stream_enable_cache(&s, &opts->stream_cache);
struct demuxer *d = demux_open(s, params, global);
if (!d) {
params->demuxer_failed = true;
free_stream(s);
}
return d;
}
static void flush_locked(demuxer_t *demuxer)
{
for (int n = 0; n < demuxer->in->num_streams; n++)
ds_flush(demuxer->in->streams[n]->ds);
demuxer->in->warned_queue_overflow = false;
demuxer->in->eof = false;
demuxer->in->last_eof = false;
2014-08-27 22:42:45 +02:00
demuxer->in->idle = true;
demuxer->filepos = -1; // implicitly synchronized
}
// clear the packet queues
void demux_flush(demuxer_t *demuxer)
{
pthread_mutex_lock(&demuxer->in->lock);
flush_locked(demuxer);
pthread_mutex_unlock(&demuxer->in->lock);
}
int demux_seek(demuxer_t *demuxer, double seek_pts, int flags)
{
struct demux_internal *in = demuxer->in;
assert(demuxer == in->d_user);
if (!demuxer->seekable) {
MP_WARN(demuxer, "Cannot seek in this file.\n");
return 0;
}
if (seek_pts == MP_NOPTS_VALUE)
return 0;
if (!(flags & SEEK_FORWARD))
flags |= SEEK_BACKWARD;
pthread_mutex_lock(&in->lock);
MP_VERBOSE(in, "queuing seek to %f%s\n", seek_pts,
in->seeking ? " (cascade)" : "");
flush_locked(demuxer);
in->seeking = true;
in->seek_flags = flags;
in->seek_pts = seek_pts;
if (!(flags & SEEK_FACTOR))
in->seek_pts = MP_ADD_PTS(in->seek_pts, -in->ts_offset);
if (!in->threading)
execute_seek(in);
pthread_cond_signal(&in->wakeup);
pthread_mutex_unlock(&in->lock);
return 1;
}
struct sh_stream *demuxer_stream_by_demuxer_id(struct demuxer *d,
enum stream_type t, int id)
{
int num = demux_get_num_stream(d);
for (int n = 0; n < num; n++) {
struct sh_stream *s = demux_get_stream(d, n);
if (s->type == t && s->demuxer_id == id)
return s;
Make audio stream index handling saner in stream switching The handling of audio stream numbering was handled in the stream selection property was a total mess. The most important issue was confusion between values used as index for demuxer->audio_streams[] array (consistently stored in demuxer->audio->id) and values stored in sh_audio->aid and used as "-aid N" option values. Now demuxer audio switch control functions and demuxer_switch_audio() are supposed to return the new value for the "-aid" option (internal MPEG demuxers still don't; the demuxer requirement could perhaps be dropped as it can be easily calculated afterwards). That is also the value returned for the "switch_audio" property. The main changes are: - Make command.c mp_property_audio() consistently use and return the "-aid" values. Before it used that as input but the array index as output, with extra mess related to demuxer_switch_audio() return value. Don't modify the audio_id option field any more. - Make demuxer_switch_audio() always return "-aid" values (like it takes as input). There are two changes for this: picking this return value in case the demuxer doesn't support switching, and overriding demuxer return value (for internal MPEG demuxers). - Make demux_lavf return "-aid" values from DEMUXER_CTRL_SWITCH_AUDIO code. This isn't actually necessary because of the override part above. Here's some history of the relevant behavior that I looked up: * For most demuxers array index and "-aid" values are the same. At least demux_mkv, (some of?) the internal MPEG demuxers and demux_ogg have differed for a long time. demux_ogg doesn't matter because it doesn't support stream switching. * Old code seemed to assume that demuxer_switch_audio() return value was array index, but this wasn't true at least for demux_mkv. * In svn r19951 reimar mostly removed use of the return value. * In r20162 ptt added mp_property_audio(). This set the global audio_id variable (-aid option value) to the return value of demuxer_switch_audio() and treated the global as the persistent value of the property, apparently assuming that it would be set to the "-aid" value, not array index. This was false for internal MPEG. * In r30124 reimar changed the property code so that even though it still modified the option value it didn't use that as the value of the property any more; instead it incorrectly used the array index. This meant that for demux_mkv the return value didn't match -aid any more (though input still did, so setting the property and querying it didn't match as they used different value systems). * In r31129 aurel made demux_lavf changes that resulted in its -aid and array index values no longer matching either. He didn't change the return value from audio switch when changing -aid, so it now matched array index only. The latter part didn't cause additional problems from r20162 though because either choice would have been broken anyway after r30124 as long as they weren't the same value.
2010-05-22 07:19:23 +02:00
}
return NULL;
}
// Set whether the given stream should return packets.
// ref_pts is used only if the stream is enabled. Then it serves as approximate
// start pts for this stream (in the worst case it is ignored).
void demuxer_select_track(struct demuxer *demuxer, struct sh_stream *stream,
double ref_pts, bool selected)
{
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
struct demux_internal *in = demuxer->in;
pthread_mutex_lock(&in->lock);
// don't flush buffers if stream is already selected / unselected
if (stream->ds->selected != selected) {
stream->ds->selected = selected;
ds_flush(stream->ds);
in->tracks_switched = true;
stream->ds->need_refresh = selected && !in->initial_state;
if (stream->ds->need_refresh)
in->ref_pts = MP_ADD_PTS(ref_pts, -in->ts_offset);
if (in->threading) {
pthread_cond_signal(&in->wakeup);
} else {
execute_trackswitch(in);
}
2013-07-11 19:20:25 +02:00
}
demux: hack for instant stream switching This removes the delay when switching audio tracks in mkv or mp4 files. Other formats are not enabled, because it's not clear whether the demuxers fulfill the requirements listed in demux.h. (Many formats definitely do not with libavformat.) Background: The demuxer packet cache buffers a certain amount of packets. This includes only packets from selected streams. We discard packets from other streams for various reasons. This introduces a problem: switching to a different audio track introduces a delay. The delay is as big as the demuxer packet cache buffer, because while the file was read ahead to fill the packet buffer, the process of reading packets also discarded all packets from the previously not selected audio stream. Once the remaining packet buffer has been played, new audio packets are available and you hear audio again. We could probably just not discard packets from unselected streams. But this would require additional memory and CPU resources, and also it's hard to tell when packets from unused streams should be discarded (we don't want to keep them forever; it'd be a memory leak). We could also issue a player hr-seek to the current playback position, which would solve the problem in 1 line of code or so. But this can be rather slow. So what we do in this commit instead is: we just seek back to the position where our current packet buffer starts, and start demuxing from this position again. This way we can get the "past" packets for the newly selected stream. For streams which were already selected the packets are simply discarded until the previous position is reached again. That latter part is the hard part. We really want to skip packets exactly until the position where we left off previously, or we will skip packets or feed packets to the decoder twice. If we assume that the demuxer is deterministic (returns exactly the same packets after a seek to a previous position), then we can try to check whether it's the same packet as the one at the end of the packet buffer. If it is, we know that the packet after it is where we left off last time. Unfortunately, this is not very robust, and maybe it can't be made robust. Currently we use the demux_packet.pos field as unique packet ID - which works fine in some scenarios, but will break in arbitrary ways if the basic requirement to the demuxer (as listed in the demux.h additions) are broken. Thus, this is enabled only for the internal mkv demuxer and the libavformat mp4 demuxer. (libavformat mkv does not work, because the packet positions are not unique. Probably could be fixed upstream, but it's not clear whether it's a bug or a feature.)
2015-02-13 21:17:17 +01:00
pthread_mutex_unlock(&in->lock);
}
void demux_set_stream_autoselect(struct demuxer *demuxer, bool autoselect)
{
assert(!demuxer->in->threading); // laziness
demuxer->in->autoselect = autoselect;
}
bool demux_stream_is_selected(struct sh_stream *stream)
{
if (!stream)
return false;
bool r = false;
pthread_mutex_lock(&stream->ds->in->lock);
r = stream->ds->selected;
pthread_mutex_unlock(&stream->ds->in->lock);
return r;
}
int demuxer_add_attachment(demuxer_t *demuxer, char *name, char *type,
void *data, size_t data_size)
{
if (!(demuxer->num_attachments % 32))
demuxer->attachments = talloc_realloc(demuxer, demuxer->attachments,
struct demux_attachment,
demuxer->num_attachments + 32);
struct demux_attachment *att = &demuxer->attachments[demuxer->num_attachments];
att->name = talloc_strdup(demuxer->attachments, name);
att->type = talloc_strdup(demuxer->attachments, type);
att->data = talloc_memdup(demuxer->attachments, data, data_size);
att->data_size = data_size;
return demuxer->num_attachments++;
}
static int chapter_compare(const void *p1, const void *p2)
{
struct demux_chapter *c1 = (void *)p1;
struct demux_chapter *c2 = (void *)p2;
if (c1->pts > c2->pts)
return 1;
else if (c1->pts < c2->pts)
return -1;
return c1->original_index > c2->original_index ? 1 :-1; // never equal
}
static void demuxer_sort_chapters(demuxer_t *demuxer)
{
qsort(demuxer->chapters, demuxer->num_chapters,
sizeof(struct demux_chapter), chapter_compare);
}
int demuxer_add_chapter(demuxer_t *demuxer, char *name,
double pts, uint64_t demuxer_id)
{
struct demux_chapter new = {
.original_index = demuxer->num_chapters,
.pts = pts,
.metadata = talloc_zero(demuxer, struct mp_tags),
.demuxer_id = demuxer_id,
};
mp_tags_set_str(new.metadata, "TITLE", name);
MP_TARRAY_APPEND(demuxer, demuxer->chapters, demuxer->num_chapters, new);
return demuxer->num_chapters - 1;
}
double demuxer_get_time_length(struct demuxer *demuxer)
{
double len;
if (demux_control(demuxer, DEMUXER_CTRL_GET_TIME_LENGTH, &len) > 0)
return len;
return -1;
}
// must be called not locked
static void update_cache(struct demux_internal *in)
{
struct demuxer *demuxer = in->d_thread;
struct stream *stream = demuxer->stream;
// Don't lock while querying the stream.
double time_length = -1;
struct mp_tags *stream_metadata = NULL;
struct stream_cache_info stream_cache_info = {.size = -1};
if (demuxer->desc->control) {
demuxer->desc->control(demuxer, DEMUXER_CTRL_GET_TIME_LENGTH,
&time_length);
}
int64_t stream_size = stream_get_size(stream);
stream_control(stream, STREAM_CTRL_GET_METADATA, &stream_metadata);
stream_control(stream, STREAM_CTRL_GET_CACHE_INFO, &stream_cache_info);
pthread_mutex_lock(&in->lock);
in->time_length = time_length;
in->stream_size = stream_size;
in->stream_cache_info = stream_cache_info;
if (stream_metadata) {
talloc_free(in->stream_metadata);
in->stream_metadata = talloc_steal(in, stream_metadata);
in->d_buffer->events |= DEMUX_EVENT_METADATA;
}
pthread_mutex_unlock(&in->lock);
}
// must be called locked
static int cached_stream_control(struct demux_internal *in, int cmd, void *arg)
{
// If the cache is active, wake up the thread to possibly update cache state.
if (in->stream_cache_info.size >= 0) {
in->force_cache_update = true;
pthread_cond_signal(&in->wakeup);
}
switch (cmd) {
case STREAM_CTRL_GET_CACHE_INFO:
if (in->stream_cache_info.size < 0)
return STREAM_UNSUPPORTED;
*(struct stream_cache_info *)arg = in->stream_cache_info;
return STREAM_OK;
case STREAM_CTRL_GET_SIZE:
if (in->stream_size < 0)
return STREAM_UNSUPPORTED;
*(int64_t *)arg = in->stream_size;
return STREAM_OK;
case STREAM_CTRL_GET_BASE_FILENAME:
if (!in->stream_base_filename)
return STREAM_UNSUPPORTED;
*(char **)arg = talloc_strdup(NULL, in->stream_base_filename);
return STREAM_OK;
}
return STREAM_ERROR;
}
// must be called locked
static int cached_demux_control(struct demux_internal *in, int cmd, void *arg)
{
switch (cmd) {
case DEMUXER_CTRL_GET_TIME_LENGTH:
if (in->time_length < 0)
return DEMUXER_CTRL_NOTIMPL;
*(double *)arg = in->time_length;
return DEMUXER_CTRL_OK;
case DEMUXER_CTRL_STREAM_CTRL: {
struct demux_ctrl_stream_ctrl *c = arg;
int r = cached_stream_control(in, c->ctrl, c->arg);
if (r == STREAM_ERROR)
break;
c->res = r;
return DEMUXER_CTRL_OK;
}
case DEMUXER_CTRL_GET_BITRATE_STATS: {
double *rates = arg;
for (int n = 0; n < STREAM_TYPE_COUNT; n++)
rates[n] = -1;
for (int n = 0; n < in->num_streams; n++) {
struct demux_stream *ds = in->streams[n]->ds;
if (ds->selected && ds->bitrate >= 0)
rates[ds->type] = MPMAX(0, rates[ds->type]) + ds->bitrate;
}
return DEMUXER_CTRL_OK;
}
case DEMUXER_CTRL_GET_READER_STATE: {
struct demux_ctrl_reader_state *r = arg;
*r = (struct demux_ctrl_reader_state){
.eof = in->last_eof,
.ts_range = {MP_NOPTS_VALUE, MP_NOPTS_VALUE},
.ts_duration = -1,
};
int num_packets = 0;
for (int n = 0; n < in->num_streams; n++) {
struct demux_stream *ds = in->streams[n]->ds;
if (ds->active) {
r->underrun |= !ds->head && !ds->eof;
r->ts_range[0] = MP_PTS_MAX(r->ts_range[0], ds->base_ts);
r->ts_range[1] = MP_PTS_MIN(r->ts_range[1], ds->last_ts);
num_packets += ds->packs;
}
}
2014-08-27 22:42:45 +02:00
r->idle = (in->idle && !r->underrun) || r->eof;
r->underrun &= !r->idle;
if (r->ts_range[0] != MP_NOPTS_VALUE && r->ts_range[1] != MP_NOPTS_VALUE)
r->ts_duration = MPMAX(0, r->ts_range[1] - r->ts_range[0]);
if (!num_packets || in->seeking)
r->ts_duration = 0;
r->ts_range[0] = MP_ADD_PTS(r->ts_range[0], in->ts_offset);
r->ts_range[1] = MP_ADD_PTS(r->ts_range[1], in->ts_offset);
return DEMUXER_CTRL_OK;
}
}
return DEMUXER_CTRL_DONTKNOW;
}
struct demux_control_args {
struct demuxer *demuxer;
int cmd;
void *arg;
int *r;
};
static void thread_demux_control(void *p)
{
struct demux_control_args *args = p;
struct demuxer *demuxer = args->demuxer;
int cmd = args->cmd;
void *arg = args->arg;
struct demux_internal *in = demuxer->in;
int r = DEMUXER_CTRL_NOTIMPL;
if (cmd == DEMUXER_CTRL_STREAM_CTRL) {
struct demux_ctrl_stream_ctrl *c = arg;
if (in->threading)
MP_VERBOSE(demuxer, "blocking for STREAM_CTRL %d\n", c->ctrl);
c->res = stream_control(demuxer->stream, c->ctrl, c->arg);
if (c->res != STREAM_UNSUPPORTED)
r = DEMUXER_CTRL_OK;
}
if (r != DEMUXER_CTRL_OK) {
if (in->threading)
MP_VERBOSE(demuxer, "blocking for DEMUXER_CTRL %d\n", cmd);
if (demuxer->desc->control)
r = demuxer->desc->control(demuxer->in->d_thread, cmd, arg);
}
*args->r = r;
}
int demux_control(demuxer_t *demuxer, int cmd, void *arg)
{
struct demux_internal *in = demuxer->in;
if (in->threading) {
pthread_mutex_lock(&in->lock);
int cr = cached_demux_control(in, cmd, arg);
pthread_mutex_unlock(&in->lock);
if (cr != DEMUXER_CTRL_DONTKNOW)
return cr;
}
int r = 0;
struct demux_control_args args = {demuxer, cmd, arg, &r};
demux_run_on_thread(demuxer, thread_demux_control, &args);
return r;
}
int demux_stream_control(demuxer_t *demuxer, int ctrl, void *arg)
{
struct demux_ctrl_stream_ctrl c = {ctrl, arg, STREAM_UNSUPPORTED};
demux_control(demuxer, DEMUXER_CTRL_STREAM_CTRL, &c);
return c.res;
}
void demux_run_on_thread(struct demuxer *demuxer, void (*fn)(void *), void *ctx)
{
struct demux_internal *in = demuxer->in;
assert(demuxer == in->d_user);
if (in->threading) {
MP_VERBOSE(in, "blocking on demuxer thread\n");
pthread_mutex_lock(&in->lock);
while (in->run_fn)
pthread_cond_wait(&in->wakeup, &in->lock);
in->run_fn = fn;
in->run_fn_arg = ctx;
pthread_cond_signal(&in->wakeup);
while (in->run_fn)
pthread_cond_wait(&in->wakeup, &in->lock);
pthread_mutex_unlock(&in->lock);
} else {
fn(ctx);
}
}
bool demux_cancel_test(struct demuxer *demuxer)
{
return mp_cancel_test(demuxer->stream->cancel);
}
struct demux_chapter *demux_copy_chapter_data(struct demux_chapter *c, int num)
{
struct demux_chapter *new = talloc_array(NULL, struct demux_chapter, num);
for (int n = 0; n < num; n++) {
new[n] = c[n];
new[n].metadata = mp_tags_dup(new, new[n].metadata);
}
return new;
}