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mpv/player/audio.c

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Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
/*
Update license headers Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 07:36:54 +00:00
* This file is part of mpv.
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
*
Update license headers Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 07:36:54 +00:00
* mpv is free software; you can redistribute it and/or modify
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
* 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.
*
Update license headers Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 07:36:54 +00:00
* mpv is distributed in the hope that it will be useful,
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
* 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
Update license headers Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 07:36:54 +00:00
* with mpv. If not, see <http://www.gnu.org/licenses/>.
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
*/
#include <stddef.h>
#include <stdbool.h>
#include <inttypes.h>
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
#include <limits.h>
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
#include <math.h>
#include <assert.h>
#include "config.h"
mpv_talloc.h: rename from talloc.h This change helps avoiding conflict with talloc.h from libtalloc.
2016-01-11 18:03:40 +00:00
#include "mpv_talloc.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
Split mpvcore/ into common/, misc/, bstr/
2013-12-17 01:39:45 +00:00
#include "common/msg.h"
encode: don't access ao->pts This field will be moved out of the ao struct. The encoding code was basically using an invalid way of accessing this field. Since the AO will be moved into its own thread too and will do its own buffering, the AO and the playback core might not even agree which sample a PTS timestamp belongs to. Add some extrapolation code to handle this case.
2014-03-07 14:23:03 +00:00
#include "common/encode.h"
Move options/config related files from mpvcore/ to options/ Since m_option.h and options.h are extremely often included, a lot of files have to be changed. Moving path.c/h to options/ is a bit questionable, but since this is mainly about access to config files (which are also handled in options/), it's probably ok.
2013-12-17 01:02:25 +00:00
#include "options/options.h"
Split mpvcore/ into common/, misc/, bstr/
2013-12-17 01:39:45 +00:00
#include "common/common.h"
audio: add AO deviation logging Pretty dumb (and doesn't handle pausing or other discontinuities), but at least somewhat idiot-proof.
2015-10-08 21:08:20 +00:00
#include "osdep/timer.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
audio: switch output to mp_audio_buffer Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
2013-11-10 22:38:18 +00:00
#include "audio/audio.h"
#include "audio/audio_buffer.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
#include "audio/decode/dec_audio.h"
#include "audio/filter/af.h"
#include "audio/out/ao.h"
#include "demux/demux.h"
video: move decoder context from sh_video into new struct This is similar to the sh_audio commit. This is mostly cosmetic in nature, except that it also adds automatical freeing of the decoder driver's state struct (which was in sh_video->context, now in dec_video->priv). Also remove all the stheader.h fields that are not needed anymore.
2013-11-23 20:36:20 +00:00
#include "video/decode/dec_video.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
Rename mp_core.h to core.h Get rid of the mp_ prefix.
2013-12-17 00:08:53 +00:00
#include "core.h"
client API: add events for video and audio reconfig
2014-02-17 01:52:26 +00:00
#include "command.h"
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
enum {
AD_OK = 0,
AD_ERR = -1,
AD_EOF = -2,
AD_NEW_FMT = -3,
AD_WAIT = -4,
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
AD_NO_PROGRESS = -5,
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
};
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
// Use pitch correction only for speed adjustments by the user, not minor sync
// correction ones.
static int get_speed_method(struct MPContext *mpctx)
{
return mpctx->opts->pitch_correction && mpctx->opts->playback_speed != 1.0
? AF_CONTROL_SET_PLAYBACK_SPEED : AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE;
}
// Try to reuse the existing filters to change playback speed. If it works,
// return true; if filter recreation is needed, return false.
static bool update_speed_filters(struct MPContext *mpctx)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
{
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct af_stream *afs = mpctx->ao_chain->af;
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
double speed = mpctx->audio_speed;
audio: use the decoder buffer's format, not sh_audio When the decoder detects a format change, it overwrites the values stored in sh_audio (this affects the members sample_format, samplerate, channels). In the case when the old audio data still needs to be played/filtered, the audio format as identified by sh_audio and the format used for the decoder buffer can mismatch. In particular, they will mismatch in the very unlikely but possible case the audio chain is reinitialized while old data is draining during a format change. Or in other words, sh_audio might contain the new format, while the audio chain is still configured to use the old format. Currently, the audio code (player/audio.c and init_audio_filters) access sh_audio to get the current format. This is in theory incorrect for the reasons mentioned above. Use the decoder buffer's format instead, which should be correct at any point.
2013-11-18 12:57:17 +00:00
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
if (afs->initialized < 1)
return false;
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 19:14:42 +00:00
// Make sure only exactly one filter changes speed; resetting them all
// and setting 1 filter is the easiest way to achieve this.
af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &(double){1});
af_control_all(afs, AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE, &(double){1});
if (speed == 1.0)
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
return !af_find_by_label(afs, "playback-speed");
audio: change how filters are inserted on playback speed changes Use a pseudo-filter when changing speed with resampling, instead of somehow changing a samplerate somewhere. This uses the same underlying mechanism, but is a bit more structured and cleaner. It also makes some of the following changes easier. Since we now always use filters to change audio speed, move most of the work set_playback_speed() does to recreate_audio_filters().
2014-11-10 19:16:25 +00:00
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 19:14:42 +00:00
// Compatibility: if the user uses --af=scaletempo, always use this
// filter to change speed. Don't insert a second filter (any) either.
if (!af_find_by_label(afs, "playback-speed") &&
af_control_any_rev(afs, AF_CONTROL_SET_PLAYBACK_SPEED, &speed))
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
return true;
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 19:14:42 +00:00
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
return !!af_control_any_rev(afs, get_speed_method(mpctx), &speed);
}
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 19:14:42 +00:00
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
// Update speed, and insert/remove filters if necessary.
static void recreate_speed_filters(struct MPContext *mpctx)
{
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct af_stream *afs = mpctx->ao_chain->af;
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
if (update_speed_filters(mpctx))
return;
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 19:14:42 +00:00
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
if (af_remove_by_label(afs, "playback-speed") < 0)
goto fail;
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
if (mpctx->audio_speed == 1.0)
return;
int method = get_speed_method(mpctx);
char *filter = method == AF_CONTROL_SET_PLAYBACK_SPEED
? "scaletempo" : "lavrresample";
if (!af_add(afs, filter, "playback-speed", NULL))
goto fail;
if (!update_speed_filters(mpctx))
goto fail;
return;
fail:
mpctx->opts->playback_speed = 1.0;
mpctx->speed_factor_a = 1.0;
mpctx->audio_speed = 1.0;
mp_notify(mpctx, MP_EVENT_CHANGE_ALL, NULL);
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 19:14:42 +00:00
}
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 19:45:50 +00:00
// Convert to gain value from dB. input <= -200dB will become 0 gain.
// Copyright (C)2002 Anders Johansson
static float from_dB(float in, float k, float mi, float ma)
{
if (in <= -200)
return 0.0;
return pow(10.0, MPCLAMP(in, mi, ma) / k);
}
static float compute_replaygain(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct ao_chain *ao_c = mpctx->ao_chain;
float rgain = 1.0;
struct replaygain_data *rg = ao_c->af->replaygain_data;
audio: merge --replaygain-track and --replaygain-album into one option This is probably better than separate options. For example, the user does not have to guess which one is applied if both options are enabled.
2017-04-26 22:21:17 +00:00
if (opts->rgain_mode && rg) {
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 19:45:50 +00:00
MP_VERBOSE(mpctx, "Replaygain: Track=%f/%f Album=%f/%f\n",
rg->track_gain, rg->track_peak,
rg->album_gain, rg->album_peak);
float gain, peak;
audio: merge --replaygain-track and --replaygain-album into one option This is probably better than separate options. For example, the user does not have to guess which one is applied if both options are enabled.
2017-04-26 22:21:17 +00:00
if (opts->rgain_mode == 1) {
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 19:45:50 +00:00
gain = rg->track_gain;
peak = rg->track_peak;
} else {
gain = rg->album_gain;
peak = rg->album_peak;
}
gain += opts->rgain_preamp;
rgain = from_dB(gain, 20.0, -200.0, 60.0);
MP_VERBOSE(mpctx, "Applying replay-gain: %f\n", rgain);
if (!opts->rgain_clip) { // clipping prevention
rgain = MPMIN(rgain, 1.0 / peak);
MP_VERBOSE(mpctx, "...with clipping prevention: %f\n", rgain);
}
} else if (opts->rgain_fallback) {
rgain = from_dB(opts->rgain_fallback, 20.0, -200.0, 60.0);
MP_VERBOSE(mpctx, "Applying fallback gain: %f\n", rgain);
}
return rgain;
}
audio: refactor mixer code and delete mixer.c mixer.c didn't really deserve to be separate anymore, as half of its contents were unnecessary glue code after recent changes. It also created a weird split between audio.c and af.c due to the fact that mixer.c could insert audio filters. With the code being in audio.c directly, together with other code that unserts filters during runtime, it will be possible to cleanup this code a bit and make it work like the video filter code. As part of this change, make the balance code work like the volume code, and add an option to back the current balance value. Also, since the balance semantics are unexpected for most users (panning between the audio channels, instead of just changing the relative volume), and there are some other volumes, formally deprecate both the old property and the new option.
2016-07-17 17:21:28 +00:00
// Called when opts->softvol_volume or opts->softvol_mute were changed.
void audio_update_volume(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c || ao_c->af->initialized < 1)
return;
float gain = MPMAX(opts->softvol_volume / 100.0, 0);
audio: fix replaygain volume scale The new replaygain code accidentally applied the linear gain as cubic volume level. Fix this by moving the computation of the volume scale out of the af_volume filter. (Still haven't verified whether the replaygain code works correctly.)
2017-04-26 22:15:22 +00:00
gain = pow(gain, 3);
audio: move replaygain control to top-level options af_volume is deprecated, and so are its replaygain sub-options. To make it possible to use replaygain without deprecated options (and of course to make it available at all after af_volume is dropped), reintroduce them as top-level options. This also means that they are easily changeable at runtime by using them as properties. Change the "volume" property to use the new update mechanism as well. We don't actually bother sharing the implementation between new and deprecated mechanisms, as the deprecated one will simply be deleted. For the from_dB() functions, we mention anders' copyright, although I'm not sure if a mere formula is copyrightable. This will have to be determined later. This whole change is mostly untested. Our distributed human CI will take care of it.
2017-04-26 19:45:50 +00:00
gain *= compute_replaygain(mpctx);
audio: refactor mixer code and delete mixer.c mixer.c didn't really deserve to be separate anymore, as half of its contents were unnecessary glue code after recent changes. It also created a weird split between audio.c and af.c due to the fact that mixer.c could insert audio filters. With the code being in audio.c directly, together with other code that unserts filters during runtime, it will be possible to cleanup this code a bit and make it work like the video filter code. As part of this change, make the balance code work like the volume code, and add an option to back the current balance value. Also, since the balance semantics are unexpected for most users (panning between the audio channels, instead of just changing the relative volume), and there are some other volumes, formally deprecate both the old property and the new option.
2016-07-17 17:21:28 +00:00
if (opts->softvol_mute == 1)
gain = 0.0;
if (!af_control_any_rev(ao_c->af, AF_CONTROL_SET_VOLUME, &gain)) {
if (gain == 1.0)
return;
MP_VERBOSE(mpctx, "Inserting volume filter.\n");
audio: deprecate most audio filters Well, ok, only 4 filters. The rest will survive in one or the other form.
2017-04-04 13:04:07 +00:00
char *args[] = {"warn", "no", NULL};
if (!(af_add(ao_c->af, "volume", "softvol", args)
audio: refactor mixer code and delete mixer.c mixer.c didn't really deserve to be separate anymore, as half of its contents were unnecessary glue code after recent changes. It also created a weird split between audio.c and af.c due to the fact that mixer.c could insert audio filters. With the code being in audio.c directly, together with other code that unserts filters during runtime, it will be possible to cleanup this code a bit and make it work like the video filter code. As part of this change, make the balance code work like the volume code, and add an option to back the current balance value. Also, since the balance semantics are unexpected for most users (panning between the audio channels, instead of just changing the relative volume), and there are some other volumes, formally deprecate both the old property and the new option.
2016-07-17 17:21:28 +00:00
&& af_control_any_rev(ao_c->af, AF_CONTROL_SET_VOLUME, &gain)))
MP_ERR(mpctx, "No volume control available.\n");
}
}
/* NOTE: Currently the balance code is seriously buggy: it always changes
* the af_pan mapping between the first two input channels and first two
* output channels to particular values. These values make sense for an
* af_pan instance that was automatically inserted for balance control
* only and is otherwise an identity transform, but if the filter was
* there for another reason, then ignoring and overriding the original
* values is completely wrong.
*/
void audio_update_balance(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c || ao_c->af->initialized < 1)
return;
float val = opts->balance;
if (af_control_any_rev(ao_c->af, AF_CONTROL_SET_PAN_BALANCE, &val))
return;
if (val == 0)
return;
struct af_instance *af_pan_balance;
if (!(af_pan_balance = af_add(ao_c->af, "pan", "autopan", NULL))) {
MP_ERR(mpctx, "No balance control available.\n");
return;
}
/* make all other channels pass through since by default pan blocks all */
for (int i = 2; i < AF_NCH; i++) {
float level[AF_NCH] = {0};
level[i] = 1.f;
af_control_ext_t arg_ext = { .ch = i, .arg = level };
af_pan_balance->control(af_pan_balance, AF_CONTROL_SET_PAN_LEVEL,
&arg_ext);
}
af_pan_balance->control(af_pan_balance, AF_CONTROL_SET_PAN_BALANCE, &val);
}
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 19:14:42 +00:00
static int recreate_audio_filters(struct MPContext *mpctx)
{
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
assert(mpctx->ao_chain);
audio: make changing playback speed slightly more robust Always use af_scaletempo if it's inserted, even if the option --audio-pitch-correction=no is set. Make sure all filters are reset on speed change. It's conceivable that dynamic changes to the filter chain at runtime leave filters around without resetting their speed parameters. Also move the code to a separate function.
2015-03-07 19:14:42 +00:00
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct af_stream *afs = mpctx->ao_chain->af;
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
if (afs->initialized < 1 && af_init(afs) < 0)
goto fail;
recreate_speed_filters(mpctx);
if (afs->initialized < 1 && af_init(afs) < 0)
goto fail;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
audio: refactor mixer code and delete mixer.c mixer.c didn't really deserve to be separate anymore, as half of its contents were unnecessary glue code after recent changes. It also created a weird split between audio.c and af.c due to the fact that mixer.c could insert audio filters. With the code being in audio.c directly, together with other code that unserts filters during runtime, it will be possible to cleanup this code a bit and make it work like the video filter code. As part of this change, make the balance code work like the volume code, and add an option to back the current balance value. Also, since the balance semantics are unexpected for most users (panning between the audio channels, instead of just changing the relative volume), and there are some other volumes, formally deprecate both the old property and the new option.
2016-07-17 17:21:28 +00:00
if (mpctx->opts->softvol == SOFTVOL_NO)
MP_ERR(mpctx, "--softvol=no is not supported anymore.\n");
audio_update_volume(mpctx);
audio_update_balance(mpctx);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: add missing audio reconfig events This also takes care of sending the required property change notifications. Fixes #2929 and maybe fixes #2920.
2016-03-13 14:52:17 +00:00
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
return 0;
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
fail:
MP_ERR(mpctx, "Couldn't find matching filter/ao format!\n");
return -1;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
int reinit_audio_filters(struct MPContext *mpctx)
{
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c)
command: allow changing filters before video chain initialization Apparently this is more intuitive. Somewhat tricky, because of the odd state after loading a file but before initializing the VO.
2014-03-30 17:21:54 +00:00
return 0;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: force refresh seek when changing audio filters Unfortunately I see no better solution. The refresh seek is skipped if the amount of buffered audio is not overly huge. Unfortunately softvol af_volume insertion still can cause this issue, because it's outside of the normal dynamic filter chain changing code. Move the video refresh call to reinit_video_filters() to make it more uniform along with the audio code.
2016-02-09 21:19:01 +00:00
double delay = 0;
if (ao_c->af->initialized > 0)
delay = af_calc_delay(ao_c->af);
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
af_uninit(ao_c->af);
player: force refresh seek when changing audio filters Unfortunately I see no better solution. The refresh seek is skipped if the amount of buffered audio is not overly huge. Unfortunately softvol af_volume insertion still can cause this issue, because it's outside of the normal dynamic filter chain changing code. Move the video refresh call to reinit_video_filters() to make it more uniform along with the audio code.
2016-02-09 21:19:01 +00:00
if (recreate_audio_filters(mpctx) < 0)
return -1;
// Only force refresh if the amount of dropped buffered data is going to
// cause "issues" for the A/V sync logic.
if (mpctx->audio_status == STATUS_PLAYING &&
mpctx->playback_pts != MP_NOPTS_VALUE && delay > 0.2)
{
queue_seek(mpctx, MPSEEK_ABSOLUTE, mpctx->playback_pts,
player: allow passing flags to queue_seek() Change the last parameter from a bool to an int, which is supposed to take bit-flags. The at this point only flag is MPSEEK_FLAG_DELAY, which replaces the previous bool parameter. The old false parameter becomes 0, the old true parameter becomes MPSEEK_FLAG_DELAY. Since the old "immediate" parameter is now essentially inverted, two coalesced immediate and delayed seeks end up as delayed instead of immediate. This change doesn't matter, since there are no relative immediate seeks anyway.
2016-08-15 19:07:32 +00:00
MPSEEK_EXACT, 0);
player: force refresh seek when changing audio filters Unfortunately I see no better solution. The refresh seek is skipped if the amount of buffered audio is not overly huge. Unfortunately softvol af_volume insertion still can cause this issue, because it's outside of the normal dynamic filter chain changing code. Move the video refresh call to reinit_video_filters() to make it more uniform along with the audio code.
2016-02-09 21:19:01 +00:00
}
return 1;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
// Call this if opts->playback_speed or mpctx->speed_factor_* change.
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
void update_playback_speed(struct MPContext *mpctx)
command: move setting playback speed to a separate function
2014-10-02 00:49:05 +00:00
{
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
mpctx->audio_speed = mpctx->opts->playback_speed * mpctx->speed_factor_a;
mpctx->video_speed = mpctx->opts->playback_speed * mpctx->speed_factor_v;
command: move setting playback speed to a separate function
2014-10-02 00:49:05 +00:00
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
if (!mpctx->ao_chain || mpctx->ao_chain->af->initialized < 1)
audio: enable pitch correction by default when playing fast Apparently this is what users want. When playing with normal speed, nothing is done. When playing slower than normal, resampling is used instead, because scaletempo (which does the pitch correction) adds too many artifacts.
2014-10-02 00:58:52 +00:00
return;
audio: do not require full audio chain reinit for speed changes Actually, it didn't really require that before (most work was avoided), but some bits had to be run anyway. Separate the speed change into a light-weight function, which merely updates already created filters, and a heavy-weight one which messes with filter insertion. This also happens to fix the case where the filters would "forget" the current speed (force resampling, change speed, hit a volume control to force af_volume insertion - it will reset speed and desync). Since we now always run the light-weight function, remove the af_scaletempo verbose message that is printed on speed setting. Other than that, all setters are cheap.
2015-11-04 20:31:53 +00:00
if (!update_speed_filters(mpctx))
recreate_audio_filters(mpctx);
command: move setting playback speed to a separate function
2014-10-02 00:49:05 +00:00
}
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
static void ao_chain_reset_state(struct ao_chain *ao_c)
{
ao_c->pts = MP_NOPTS_VALUE;
audio: move pts reset check Reduces the dependency of the filter/output code on the decoder.
2016-01-29 21:44:20 +00:00
ao_c->pts_reset = false;
audio: release pending audio frame on seeking
2016-01-25 12:41:28 +00:00
talloc_free(ao_c->input_frame);
ao_c->input_frame = NULL;
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
af_seek_reset(ao_c->af);
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
mp_audio_buffer_clear(ao_c->ao_buffer);
player: refactor: some more minor decoder/output decoupling These changes don't make too much sense without context, but are preparation for later. Then the audio_src/video_src fields will be actually be NULL under circumstances.
2016-01-29 21:46:28 +00:00
if (ao_c->audio_src)
audio_reset_decoding(ao_c->audio_src);
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
}
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
void reset_audio_state(struct MPContext *mpctx)
{
player: refactor: some more minor decoder/output decoupling These changes don't make too much sense without context, but are preparation for later. Then the audio_src/video_src fields will be actually be NULL under circumstances.
2016-01-29 21:46:28 +00:00
if (mpctx->ao_chain)
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
ao_chain_reset_state(mpctx->ao_chain);
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
mpctx->audio_status = mpctx->ao_chain ? STATUS_SYNCING : STATUS_EOF;
player: minor simplification in A/V-sync related code Just minor things.
2015-01-30 22:49:30 +00:00
mpctx->delay = 0;
player: limit speed change in display-sync adrop mode Discontinuities (like toggling fullscreen) can cause multiple frames to be dropped in succession, which sounds very weird. It's better to drop some video frames instead to compensate for larger desyncs. We roughly base it on the maximum allowed speed changes (audio change is "additional" to the video change to account for deviations when playing at max. video speed change).
2015-11-03 19:29:25 +00:00
mpctx->audio_drop_throttle = 0;
audio: add AO deviation logging Pretty dumb (and doesn't handle pausing or other discontinuities), but at least somewhat idiot-proof.
2015-10-08 21:08:20 +00:00
mpctx->audio_stat_start = 0;
player: gross hack to improve non-hr seeking with external audio tracks Relative seeks backwards with external audio tracks does not always work well: it tends to happen that video seek back further than audio, so audio will remain silent until the audio's after-seek position is reached. This happens because we strictly seek both video and audio demuxer to the approximate desirted target PTS, and then start decoding from that. Commit 81358380 removes an older method that was supposed to deal with this. It was sort of bad, because it could lead to the playback core freezing by waiting on network. Ideally, the demuxer layer would probably somehow deal with such seeks, and do them in a way the audio is seeked after video. Currently this is infeasible, because the demuxer layer assumes a single demuxer, and external tracks simply use separate demuxer layers. (MPlayer actually had a pseudo-demuxer that joined external tracks into a single demuxer, but this is not flexible enough - and also, the demuxer layer as it currently exists can't deal with dynamically removing external tracks either. Maybe some time in the future.) Instead, add a gross hack, that essentially reseeks the audio if it detects that it's too far off. The result is actually not too bad, because we can reuse the mechanism that is used for instant track switching. This way we can make sure of the right position, without having to care about certain other issues. It should be noted that if the audio demuxer is used for other tracks too, and the demuxer does not support refresh seeking, audio will probably be off by even a higher amount. But this should be rare.
2016-08-07 14:29:13 +00:00
mpctx->audio_allow_second_chance_seek = false;
player: split seek_reset() This also reduces some code duplication with other parts of the code. The changfe is mostly cosmetic, although there are also some subtle changes in behavior. At least one change is that the big desync message is now printed after every seek.
2014-07-30 21:01:55 +00:00
}
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
void uninit_audio_out(struct MPContext *mpctx)
{
if (mpctx->ao) {
// Note: with gapless_audio, stop_play is not correctly set
if (mpctx->opts->gapless_audio || mpctx->stop_play == AT_END_OF_FILE)
ao_drain(mpctx->ao);
ao_uninit(mpctx->ao);
audio: add some change notifications We must be sure that every change comes with a notification. Otherwise, some property changes could possibly be missed.
2015-06-09 16:29:11 +00:00
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
}
mpctx->ao = NULL;
talloc_free(mpctx->ao_decoder_fmt);
mpctx->ao_decoder_fmt = NULL;
}
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
static void ao_chain_uninit(struct ao_chain *ao_c)
{
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
struct track *track = ao_c->track;
if (track) {
assert(track->ao_c == ao_c);
track->ao_c = NULL;
assert(track->d_audio == ao_c->audio_src);
track->d_audio = NULL;
audio_uninit(ao_c->audio_src);
}
if (ao_c->filter_src)
lavfi_set_connected(ao_c->filter_src, false);
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
af_destroy(ao_c->af);
player: free queued audio frame on uninit
2016-01-23 10:41:19 +00:00
talloc_free(ao_c->input_frame);
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
talloc_free(ao_c->ao_buffer);
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
talloc_free(ao_c);
}
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
void uninit_audio_chain(struct MPContext *mpctx)
{
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
if (mpctx->ao_chain) {
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
ao_chain_uninit(mpctx->ao_chain);
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
mpctx->ao_chain = NULL;
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
mpctx->audio_status = STATUS_EOF;
audio: add some change notifications We must be sure that every change comes with a notification. Otherwise, some property changes could possibly be missed.
2015-06-09 16:29:11 +00:00
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
}
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
static void reinit_audio_filters_and_output(struct MPContext *mpctx)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
{
struct MPOpts *opts = mpctx->opts;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct ao_chain *ao_c = mpctx->ao_chain;
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
assert(ao_c);
player: remove some further current_track dependencies Now it's used for initialization only for audio and video.
2016-02-05 22:41:44 +00:00
struct track *track = ao_c->track;
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
struct af_stream *afs = ao_c->af;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: refactor: some more minor decoder/output decoupling These changes don't make too much sense without context, but are preparation for later. Then the audio_src/video_src fields will be actually be NULL under circumstances.
2016-01-29 21:46:28 +00:00
if (ao_c->input_frame)
mp_audio_copy_config(&ao_c->input_format, ao_c->input_frame);
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct mp_audio in_format = ao_c->input_format;
audio: use the decoder buffer's format, not sh_audio When the decoder detects a format change, it overwrites the values stored in sh_audio (this affects the members sample_format, samplerate, channels). In the case when the old audio data still needs to be played/filtered, the audio format as identified by sh_audio and the format used for the decoder buffer can mismatch. In particular, they will mismatch in the very unlikely but possible case the audio chain is reinitialized while old data is draining during a format change. Or in other words, sh_audio might contain the new format, while the audio chain is still configured to use the old format. Currently, the audio code (player/audio.c and init_audio_filters) access sh_audio to get the current format. This is in theory incorrect for the reasons mentioned above. Use the decoder buffer's format instead, which should be correct at any point.
2013-11-18 12:57:17 +00:00
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
if (!mp_audio_config_valid(&in_format)) {
// We don't know the audio format yet - so configure it later as we're
// resyncing. fill_audio_buffers() will call this function again.
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
return;
}
audio: enable pitch correction by default when playing fast Apparently this is what users want. When playing with normal speed, nothing is done. When playing slower than normal, resampling is used instead, because scaletempo (which does the pitch correction) adds too many artifacts.
2014-10-02 00:58:52 +00:00
// Weak gapless audio: drain AO on decoder format changes
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
if (mpctx->ao_decoder_fmt && mpctx->ao && opts->gapless_audio < 0 &&
!mp_audio_config_equals(mpctx->ao_decoder_fmt, &in_format))
audio: add a "weak" gapless mode, and make it default Basically, this allows gapless playback with similar files (including the ordered chapter case), while still being robust in general. The implementation is quite simplistic on purpose, in order to avoid all the weird corner cases that can occur when creating the filter chain. The consequence is that it might do not-gapless playback in more cases when needed, but if that bothers you, you still can use the normal gapless mode. Just using "--gapless-audio" or "--gapless-audio=yes" selects the old mode.
2014-06-08 21:54:05 +00:00
{
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
uninit_audio_out(mpctx);
audio: add a "weak" gapless mode, and make it default Basically, this allows gapless playback with similar files (including the ordered chapter case), while still being robust in general. The implementation is quite simplistic on purpose, in order to avoid all the weird corner cases that can occur when creating the filter chain. The consequence is that it might do not-gapless playback in more cases when needed, but if that bothers you, you still can use the normal gapless mode. Just using "--gapless-audio" or "--gapless-audio=yes" selects the old mode.
2014-06-08 21:54:05 +00:00
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
if (mpctx->ao && mp_audio_config_equals(&in_format, &afs->input))
return;
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
audio: don't force any parameters if spdif is used The existing code only ignored --audio-channels, but not --audio-rate or --audio-format if spdif passthrough is used. Setting these makes no sense.
2015-01-20 13:33:08 +00:00
afs->output = (struct mp_audio){0};
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
if (mpctx->ao) {
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
ao_get_format(mpctx->ao, &afs->output);
audio: fix format function consistency issues Replace all the check macros with function calls. Give them all the same case and naming schema. Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes(). Introduce af_fmt_is_pcm(), and use it in situations that used !AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format was. It simply meant "not PCM".
2015-06-26 21:06:37 +00:00
} else if (af_fmt_is_pcm(in_format.format)) {
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
afs->output.rate = opts->force_srate;
mp_audio_set_format(&afs->output, opts->audio_output_format);
audio: use --audio-channels=auto behavior, except on ALSA This commit adds an --audio-channel=auto-safe mode, and makes it the default. This mode behaves like "auto" with most AOs, except with ao_alsa. The intention is to allow multichannel output by default on sane APIs. ALSA is not sane as in it's so low level that it will e.g. configure any layout over HDMI, even if the connected A/V receiver does not support it. The HDMI fuckup is of course not ALSA's fault, but other audio APIs normally isolate applications from dealing with this and require the user to globally configure the correct output layout. This will help with other AOs too. ao_lavc (encoding) is changed to the new semantics as well, because it used to force stereo (perhaps because encoding mode is supposed to produce safe files for crap devices?). Exclusive mode output on Windows might need to be adjusted accordingly, as it grants the same kind of low level access as ALSA (requires more research). In addition to the things mentioned above, the --audio-channels option is extended to accept a set of channel layouts. This is supposed to be the correct way to configure mpv ALSA multichannel output. You need to put a list of channel layouts that your A/V receiver supports.
2016-08-04 18:49:20 +00:00
if (opts->audio_output_channels.num_chmaps == 1) {
mp_audio_set_channels(&afs->output,
&opts->audio_output_channels.chmaps[0]);
}
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
// filter input format: same as codec's output format:
audio: make decoders output refcounted frames This rewrites the audio decode loop to some degree. Audio filters don't do refcounted frames yet, so af.c contains a hacky "emulation". Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of estimating how much audio we need to filter, we always filter full frames. Maybe this should be adjusted later: in case filtering increases the volume of the audio data, we should try not to buffer too much filter output by reducing the input that is fed at once. For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it doesn't seem worth the trouble.
2014-11-10 21:01:23 +00:00
afs->input = in_format;
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
audio: cosmetics: collapse a function There's no need for build_afilter_chain() to be a separate function anymore.
2014-07-24 13:27:40 +00:00
// Determine what the filter chain outputs. recreate_audio_filters() also
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
// needs this for testing whether playback speed is changed by resampling
// or using a special filter.
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
if (af_init(afs) < 0) {
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
MP_ERR(mpctx, "Error at audio filter chain pre-init!\n");
goto init_error;
}
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
if (!mpctx->ao) {
audio: use --audio-channels=auto behavior, except on ALSA This commit adds an --audio-channel=auto-safe mode, and makes it the default. This mode behaves like "auto" with most AOs, except with ao_alsa. The intention is to allow multichannel output by default on sane APIs. ALSA is not sane as in it's so low level that it will e.g. configure any layout over HDMI, even if the connected A/V receiver does not support it. The HDMI fuckup is of course not ALSA's fault, but other audio APIs normally isolate applications from dealing with this and require the user to globally configure the correct output layout. This will help with other AOs too. ao_lavc (encoding) is changed to the new semantics as well, because it used to force stereo (perhaps because encoding mode is supposed to produce safe files for crap devices?). Exclusive mode output on Windows might need to be adjusted accordingly, as it grants the same kind of low level access as ALSA (requires more research). In addition to the things mentioned above, the --audio-channels option is extended to accept a set of channel layouts. This is supposed to be the correct way to configure mpv ALSA multichannel output. You need to put a list of channel layouts that your A/V receiver supports.
2016-08-04 18:49:20 +00:00
int ao_flags = 0;
audio: add option for falling back to ao_null The manpage entry explains this. (Maybe this option could be always enabled and removed. I don't quite remember what valid use-cases there are for just disabling audio entirely, other than that this is also needed for audio decoder init failure.)
2015-10-05 16:53:02 +00:00
bool spdif_fallback = af_fmt_is_spdif(afs->output.format) &&
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
ao_c->spdif_passthrough;
audio: use --audio-channels=auto behavior, except on ALSA This commit adds an --audio-channel=auto-safe mode, and makes it the default. This mode behaves like "auto" with most AOs, except with ao_alsa. The intention is to allow multichannel output by default on sane APIs. ALSA is not sane as in it's so low level that it will e.g. configure any layout over HDMI, even if the connected A/V receiver does not support it. The HDMI fuckup is of course not ALSA's fault, but other audio APIs normally isolate applications from dealing with this and require the user to globally configure the correct output layout. This will help with other AOs too. ao_lavc (encoding) is changed to the new semantics as well, because it used to force stereo (perhaps because encoding mode is supposed to produce safe files for crap devices?). Exclusive mode output on Windows might need to be adjusted accordingly, as it grants the same kind of low level access as ALSA (requires more research). In addition to the things mentioned above, the --audio-channels option is extended to accept a set of channel layouts. This is supposed to be the correct way to configure mpv ALSA multichannel output. You need to put a list of channel layouts that your A/V receiver supports.
2016-08-04 18:49:20 +00:00
if (opts->ao_null_fallback && !spdif_fallback)
ao_flags |= AO_INIT_NULL_FALLBACK;
player: add --audio-stream-silence Completely insane that this has to be done. Crap for compensating HDMI crap.
2016-08-09 14:22:06 +00:00
if (opts->audio_stream_silence)
ao_flags |= AO_INIT_STREAM_SILENCE;
audio/out: deprecate "exclusive" sub-options And introduce a global option which does this. Or more precisely, this deprecates the global wasapi and coreaudio options, and adds a new one that merges their functionality. (Due to the way the sub-option deprecation mechanism works, this is simpler.)
2016-09-05 19:07:34 +00:00
if (opts->audio_exclusive)
ao_flags |= AO_INIT_EXCLUSIVE;
audio: do not apply --audio-channels if spdif passthrough is in use If spdif is enabled, the channel layout has no meaning other than setting the number of channels. The number of channels must be fixed to achieve the exact bitrate required. Fixes #3445.
2016-08-22 10:12:10 +00:00
if (af_fmt_is_pcm(afs->output.format)) {
if (!opts->audio_output_channels.set ||
opts->audio_output_channels.auto_safe)
ao_flags |= AO_INIT_SAFE_MULTICHANNEL_ONLY;
mp_chmap_sel_list(&afs->output.channels,
opts->audio_output_channels.chmaps,
opts->audio_output_channels.num_chmaps);
}
audio: add option for falling back to ao_null The manpage entry explains this. (Maybe this option could be always enabled and removed. I don't quite remember what valid use-cases there are for just disabling audio entirely, other than that this is also needed for audio decoder init failure.)
2015-10-05 16:53:02 +00:00
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
mp_audio_set_channels(&afs->output, &afs->output.channels);
player, ao, vo: don't call mp_input_wakeup() directly Currently, calling mp_input_wakeup() will wake up the core thread (also called the playloop). This seems odd, but currently the core indeed calls mp_input_wait() when it has nothing more to do. It's done this way because MPlayer used input_ctx as central "mainloop". This is probably going to change. Remove direct calls to this function, and replace it with mp_wakeup_core() calls. ao and vo are changed to use opaque callbacks and not use input_ctx for this purpose. Other code already uses opaque callbacks, or has legitimate reasons to use input_ctx directly (such as sending actual user input).
2016-09-16 12:23:54 +00:00
mpctx->ao = ao_init_best(mpctx->global, ao_flags, mp_wakeup_core_cb,
mpctx, mpctx->encode_lavc_ctx, afs->output.rate,
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
afs->output.format, afs->output.channels);
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
ao_c->ao = mpctx->ao;
audio: do not allow AO to change passthrough formats This makes no sense, because the format can't be converted anyway. It just sets up the filter chain init code, which will vomit a bunch of useless and confusing messages. So uninit and fail explicitly when this happens.
2015-06-05 20:34:22 +00:00
struct mp_audio fmt = {0};
if (mpctx->ao)
ao_get_format(mpctx->ao, &fmt);
// Verify passthrough format was not changed.
audio: fix format function consistency issues Replace all the check macros with function calls. Give them all the same case and naming schema. Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes(). Introduce af_fmt_is_pcm(), and use it in situations that used !AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format was. It simply meant "not PCM".
2015-06-26 21:06:37 +00:00
if (mpctx->ao && af_fmt_is_spdif(afs->output.format)) {
audio: do not allow AO to change passthrough formats This makes no sense, because the format can't be converted anyway. It just sets up the filter chain init code, which will vomit a bunch of useless and confusing messages. So uninit and fail explicitly when this happens.
2015-06-05 20:34:22 +00:00
if (!mp_audio_config_equals(&afs->output, &fmt)) {
MP_ERR(mpctx, "Passthrough format unsupported.\n");
ao_uninit(mpctx->ao);
mpctx->ao = NULL;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
ao_c->ao = NULL;
audio: do not allow AO to change passthrough formats This makes no sense, because the format can't be converted anyway. It just sets up the filter chain init code, which will vomit a bunch of useless and confusing messages. So uninit and fail explicitly when this happens.
2015-06-05 20:34:22 +00:00
}
}
if (!mpctx->ao) {
audio: add --audio-spdif as new method for enabling passthrough This provides a new method for enabling spdif passthrough. The old method via --ad (--ad=spdif:ac3 etc.) is deprecated. The deprecated method will probably stop working at some point. This also supports PCM fallback. One caveat is that it will lose at least 1 audio packet in doing so. (I don't care enough to prevent this.) (This is named after the old S/PDIF connector, because it uses the same underlying technology as far as the higher level protoco is concerned. Also, the user should be renamed that passthrough is backwards.)
2015-06-05 20:35:43 +00:00
// If spdif was used, try to fallback to PCM.
player: refactor: some more minor decoder/output decoupling These changes don't make too much sense without context, but are preparation for later. Then the audio_src/video_src fields will be actually be NULL under circumstances.
2016-01-29 21:46:28 +00:00
if (spdif_fallback && ao_c->audio_src) {
audio: fix spdif PCM fallback With the format left untouched, this would just try to reinit with a spdif format again. We're not clearing the format in reset_audio_state() so the audio chain can be recreated any time without having to wait for a frame to be decoded.
2016-01-25 12:43:43 +00:00
MP_VERBOSE(mpctx, "Falling back to PCM output.\n");
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
ao_c->spdif_passthrough = false;
ao_c->spdif_failed = true;
player: refactor: eliminate MPContext.d_audio
2016-01-21 21:24:20 +00:00
ao_c->audio_src->try_spdif = false;
if (!audio_init_best_codec(ao_c->audio_src))
audio: add --audio-spdif as new method for enabling passthrough This provides a new method for enabling spdif passthrough. The old method via --ad (--ad=spdif:ac3 etc.) is deprecated. The deprecated method will probably stop working at some point. This also supports PCM fallback. One caveat is that it will lose at least 1 audio packet in doing so. (I don't care enough to prevent this.) (This is named after the old S/PDIF connector, because it uses the same underlying technology as far as the higher level protoco is concerned. Also, the user should be renamed that passthrough is backwards.)
2015-06-05 20:35:43 +00:00
goto init_error;
reset_audio_state(mpctx);
audio: fix spdif PCM fallback With the format left untouched, this would just try to reinit with a spdif format again. We're not clearing the format in reset_audio_state() so the audio chain can be recreated any time without having to wait for a frame to be decoded.
2016-01-25 12:43:43 +00:00
ao_c->input_format = (struct mp_audio){0};
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx); // reinit with new format next time
audio: add --audio-spdif as new method for enabling passthrough This provides a new method for enabling spdif passthrough. The old method via --ad (--ad=spdif:ac3 etc.) is deprecated. The deprecated method will probably stop working at some point. This also supports PCM fallback. One caveat is that it will lose at least 1 audio packet in doing so. (I don't care enough to prevent this.) (This is named after the old S/PDIF connector, because it uses the same underlying technology as far as the higher level protoco is concerned. Also, the user should be renamed that passthrough is backwards.)
2015-06-05 20:35:43 +00:00
return;
}
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
MP_ERR(mpctx, "Could not open/initialize audio device -> no sound.\n");
client API: better error reporting Give somewhat more information on playback failure.
2014-10-28 15:19:07 +00:00
mpctx->error_playing = MPV_ERROR_AO_INIT_FAILED;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
goto init_error;
}
audio: switch output to mp_audio_buffer Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
2013-11-10 22:38:18 +00:00
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
mp_audio_buffer_reinit(ao_c->ao_buffer, &fmt);
audio: refactor some aspects of filter chain setup There's no real reason why audio_init_filter() should exist. Just use af_init or af_reinit directly. (We lose a useless message; the same information is printed in a quite close place with more details.) Requires less code, and the way the filter chain is marked as having failed to initialize allows just switching off audio instead of crashing if trying to insert a volume filter in mixer.c fails, and recreating the old filter chain fails too.
2014-10-01 23:47:27 +00:00
afs->output = fmt;
audio: avoid one more redundant audio filter reinit Only reinit filters if it's actually needed. This is also slightly easier to understand: if you look at the code, it should now be more obvious why a reinit is needed (hopefully).
2015-04-07 11:38:03 +00:00
if (!mp_audio_config_equals(&afs->output, &afs->filter_output))
afs->initialized = 0;
audio/out: make ao struct opaque We want to move the AO to its own thread. There's no technical reason for making the ao struct opaque to do this. But it helps us sleep at night, because we can control access to shared state better.
2014-03-07 14:24:32 +00:00
audio: add a "weak" gapless mode, and make it default Basically, this allows gapless playback with similar files (including the ordered chapter case), while still being robust in general. The implementation is quite simplistic on purpose, in order to avoid all the weird corner cases that can occur when creating the filter chain. The consequence is that it might do not-gapless playback in more cases when needed, but if that bothers you, you still can use the normal gapless mode. Just using "--gapless-audio" or "--gapless-audio=yes" selects the old mode.
2014-06-08 21:54:05 +00:00
mpctx->ao_decoder_fmt = talloc(NULL, struct mp_audio);
*mpctx->ao_decoder_fmt = in_format;
audio: do not allow AO to change passthrough formats This makes no sense, because the format can't be converted anyway. It just sets up the filter chain init code, which will vomit a bunch of useless and confusing messages. So uninit and fail explicitly when this happens.
2015-06-05 20:34:22 +00:00
MP_INFO(mpctx, "AO: [%s] %s\n", ao_get_name(mpctx->ao),
audio: make mp_audio_config_to_str return a stack-allocated string Simpler overall.
2014-11-25 10:11:31 +00:00
mp_audio_config_to_str(&fmt));
audio: do not allow AO to change passthrough formats This makes no sense, because the format can't be converted anyway. It just sets up the filter chain init code, which will vomit a bunch of useless and confusing messages. So uninit and fail explicitly when this happens.
2015-06-05 20:34:22 +00:00
MP_VERBOSE(mpctx, "AO: Description: %s\n", ao_get_description(mpctx->ao));
player: set PulseAudio stream title to window title Set the PulseAudio stream title, just like the VO window title is set. Refactor update_vo_window_title() so that we can use it for AOs too. The ao_pulse.c bit is stolen from MPlayer.
2013-11-09 23:49:13 +00:00
update_window_title(mpctx, true);
player: add --audio-wait-open options Complements the option added in the previous commit.
2016-08-09 14:26:44 +00:00
ao_c->ao_resume_time =
opts->audio_wait_open > 0 ? mp_time_sec() + opts->audio_wait_open : 0;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
if (recreate_audio_filters(mpctx) < 0)
goto init_error;
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
update_playback_speed(mpctx);
audio: enable pitch correction by default when playing fast Apparently this is what users want. When playing with normal speed, nothing is done. When playing slower than normal, resampling is used instead, because scaletempo (which does the pitch correction) adds too many artifacts.
2014-10-02 00:58:52 +00:00
player: add missing audio reconfig events This also takes care of sending the required property change notifications. Fixes #2929 and maybe fixes #2920.
2016-03-13 14:52:17 +00:00
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
return;
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
init_error:
uninit_audio_chain(mpctx);
uninit_audio_out(mpctx);
player: remove some further current_track dependencies Now it's used for initialization only for audio and video.
2016-02-05 22:41:44 +00:00
error_on_track(mpctx, track);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
}
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
int init_audio_decoder(struct MPContext *mpctx, struct track *track)
{
assert(!track->d_audio);
if (!track->stream)
goto init_error;
track->d_audio = talloc_zero(NULL, struct dec_audio);
struct dec_audio *d_audio = track->d_audio;
d_audio->log = mp_log_new(d_audio, mpctx->log, "!ad");
d_audio->global = mpctx->global;
d_audio->opts = mpctx->opts;
d_audio->header = track->stream;
audio/video: expose codec info as separate field Preparation for the timeline rewrite. The codec will be able to change, the stream header not.
2016-02-15 19:34:45 +00:00
d_audio->codec = track->stream->codec;
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
d_audio->try_spdif = true;
if (!audio_init_best_codec(d_audio))
goto init_error;
return 1;
init_error:
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
if (track->sink)
lavfi_set_connected(track->sink, false);
track->sink = NULL;
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
audio_uninit(track->d_audio);
track->d_audio = NULL;
error_on_track(mpctx, track);
return 0;
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
void reinit_audio_chain(struct MPContext *mpctx)
{
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
reinit_audio_chain_src(mpctx, NULL);
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
void reinit_audio_chain_src(struct MPContext *mpctx, struct lavfi_pad *src)
{
struct track *track = NULL;
struct sh_stream *sh = NULL;
if (!src) {
track = mpctx->current_track[0][STREAM_AUDIO];
player: close audio device on no audio track Close the audio device if there is no audio track (or stream) in the current file. It will be opened again if the next file should contain audio. Signed-off-by: Marko Hauptvogel <marko.hauptvogel@googlemail.com> Signed-off-by: wm4 <wm4@nowhere>
2017-04-16 09:02:30 +00:00
if (!track) {
uninit_audio_out(mpctx);
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
return;
player: close audio device on no audio track Close the audio device if there is no audio track (or stream) in the current file. It will be opened again if the next file should contain audio. Signed-off-by: Marko Hauptvogel <marko.hauptvogel@googlemail.com> Signed-off-by: wm4 <wm4@nowhere>
2017-04-16 09:02:30 +00:00
}
player: remove dead code Fixes CID 1350055 and CID 1350054.
2016-02-12 15:05:46 +00:00
sh = track->stream;
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
if (!sh) {
uninit_audio_out(mpctx);
goto no_audio;
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
}
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
assert(!mpctx->ao_chain);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
mp_notify(mpctx, MPV_EVENT_AUDIO_RECONFIG, NULL);
player: remove double assignment in declaration Fixes CID 1350058. (Still looks like this might be valid C, in some fucked up way.)
2016-02-12 15:00:14 +00:00
struct ao_chain *ao_c = talloc_zero(NULL, struct ao_chain);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
mpctx->ao_chain = ao_c;
ao_c->log = mpctx->log;
ao_c->af = af_new(mpctx->global);
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
if (sh)
ao_c->af->replaygain_data = sh->codec->replaygain_data;
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
ao_c->spdif_passthrough = true;
ao_c->pts = MP_NOPTS_VALUE;
player: move audio and video decoder init to separate functions Preparation.
2016-02-05 20:50:37 +00:00
ao_c->ao_buffer = mp_audio_buffer_create(NULL);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
ao_c->ao = mpctx->ao;
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
ao_c->filter_src = src;
if (!ao_c->filter_src) {
ao_c->track = track;
track->ao_c = ao_c;
if (!init_audio_decoder(mpctx, track))
goto init_error;
ao_c->audio_src = track->d_audio;
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
reset_audio_state(mpctx);
if (mpctx->ao) {
struct mp_audio fmt;
ao_get_format(mpctx->ao, &fmt);
mp_audio_buffer_reinit(ao_c->ao_buffer, &fmt);
}
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
return;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
init_error:
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
uninit_audio_chain(mpctx);
uninit_audio_out(mpctx);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
no_audio:
player: remove some further current_track dependencies Now it's used for initialization only for audio and video.
2016-02-05 22:41:44 +00:00
error_on_track(mpctx, track);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
// Return pts value corresponding to the end point of audio written to the
// ao so far.
double written_audio_pts(struct MPContext *mpctx)
{
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct ao_chain *ao_c = mpctx->ao_chain;
if (!ao_c)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
return MP_NOPTS_VALUE;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct mp_audio in_format = ao_c->input_format;
audio: use the decoder buffer's format, not sh_audio When the decoder detects a format change, it overwrites the values stored in sh_audio (this affects the members sample_format, samplerate, channels). In the case when the old audio data still needs to be played/filtered, the audio format as identified by sh_audio and the format used for the decoder buffer can mismatch. In particular, they will mismatch in the very unlikely but possible case the audio chain is reinitialized while old data is draining during a format change. Or in other words, sh_audio might contain the new format, while the audio chain is still configured to use the old format. Currently, the audio code (player/audio.c and init_audio_filters) access sh_audio to get the current format. This is in theory incorrect for the reasons mentioned above. Use the decoder buffer's format instead, which should be correct at any point.
2013-11-18 12:57:17 +00:00
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
if (!mp_audio_config_valid(&in_format) || ao_c->af->initialized < 1)
cosmetics: remove a stray ';'
2014-09-04 21:25:11 +00:00
return MP_NOPTS_VALUE;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
// first calculate the end pts of audio that has been output by decoder
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
double a_pts = ao_c->pts;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
if (a_pts == MP_NOPTS_VALUE)
return MP_NOPTS_VALUE;
audio: switch output to mp_audio_buffer Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
2013-11-10 22:38:18 +00:00
// Data buffered in audio filters, measured in seconds of "missing" output
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
double buffered_output = af_calc_delay(ao_c->af);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
// Data that was ready for ao but was buffered because ao didn't fully
// accept everything to internal buffers yet
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
buffered_output += mp_audio_buffer_seconds(ao_c->ao_buffer);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
// Filters divide audio length by audio_speed, so multiply by it
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
// to get the length in original units without speedup or slowdown
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
a_pts -= buffered_output * mpctx->audio_speed;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: use demuxer ts offset to simplify timeline ts handling Use the demux_set_ts_offset() added in the previous commit to base each timeline segment to use timestamps according to its relative position within the overall timeline. As a consequence we don't need to care about these timestamps anymore, and everything becomes simpler. (Another minor but delicious nugget of sanity.)
2015-11-16 22:15:59 +00:00
return a_pts;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
// Return pts value corresponding to currently playing audio.
double playing_audio_pts(struct MPContext *mpctx)
{
double pts = written_audio_pts(mpctx);
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
if (pts == MP_NOPTS_VALUE || !mpctx->ao)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
return pts;
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
return pts - mpctx->audio_speed * ao_get_delay(mpctx->ao);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
audio: remove an unused parameter
2015-07-24 15:27:40 +00:00
static int write_to_ao(struct MPContext *mpctx, struct mp_audio *data, int flags)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
{
if (mpctx->paused)
return 0;
struct ao *ao = mpctx->ao;
audio/out: make ao struct opaque We want to move the AO to its own thread. There's no technical reason for making the ao struct opaque to do this. But it helps us sleep at night, because we can control access to shared state better.
2014-03-07 14:24:32 +00:00
struct mp_audio out_format;
ao_get_format(ao, &out_format);
encode: don't access ao->pts This field will be moved out of the ao struct. The encoding code was basically using an invalid way of accessing this field. Since the AO will be moved into its own thread too and will do its own buffering, the AO and the playback core might not even agree which sample a PTS timestamp belongs to. Add some extrapolation code to handle this case.
2014-03-07 14:23:03 +00:00
#if HAVE_ENCODING
ao_lavc: Fix design of audio pts handling. There was confusion about what should go into audio pts calculation and what not (mainly due to the audio push thread). This has been fixed by using the playing - not written - audio pts (which properly takes into account the ao's buffer), and incrementing the samples count only by the amount of samples actually taken from the buffer (unfortunately this now forces us to keep the lock too long for my taste).
2014-07-16 13:08:06 +00:00
encode_lavc_set_audio_pts(mpctx->encode_lavc_ctx, playing_audio_pts(mpctx));
encode: don't access ao->pts This field will be moved out of the ao struct. The encoding code was basically using an invalid way of accessing this field. Since the AO will be moved into its own thread too and will do its own buffering, the AO and the playback core might not even agree which sample a PTS timestamp belongs to. Add some extrapolation code to handle this case.
2014-03-07 14:23:03 +00:00
#endif
audio: don't write audio when paused This is probably "safer". Without it, we will play 1 sample, because the logic was written in a way to decode 1 sample if audio is paused. 1 sample usually will initialize the audio PTS, but not play any real audio. Also see previous commit. In ancient times, this actually used 1 byte (instead of 1 sample), so clearly no sample was written, unless the audio was 8-bit mono.
2014-03-09 00:26:24 +00:00
if (data->samples == 0)
return 0;
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
double real_samplerate = out_format.rate / mpctx->audio_speed;
audio: switch output to mp_audio_buffer Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
2013-11-10 22:38:18 +00:00
int played = ao_play(mpctx->ao, data->planes, data->samples, flags);
assert(played <= data->samples);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
if (played > 0) {
audio: switch output to mp_audio_buffer Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
2013-11-10 22:38:18 +00:00
mpctx->shown_aframes += played;
mpctx->delay += played / real_samplerate;
audio: add AO deviation logging Pretty dumb (and doesn't handle pausing or other discontinuities), but at least somewhat idiot-proof.
2015-10-08 21:08:20 +00:00
mpctx->written_audio += played / (double)out_format.rate;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
return played;
}
return 0;
}
audio: add AO deviation logging Pretty dumb (and doesn't handle pausing or other discontinuities), but at least somewhat idiot-proof.
2015-10-08 21:08:20 +00:00
static void dump_audio_stats(struct MPContext *mpctx)
{
if (!mp_msg_test(mpctx->log, MSGL_STATS))
return;
player: reset AO stats on pause and other discontinuities It's annoying.
2015-10-28 22:56:31 +00:00
if (mpctx->audio_status != STATUS_PLAYING || !mpctx->ao || mpctx->paused) {
mpctx->audio_stat_start = 0;
audio: add AO deviation logging Pretty dumb (and doesn't handle pausing or other discontinuities), but at least somewhat idiot-proof.
2015-10-08 21:08:20 +00:00
return;
player: reset AO stats on pause and other discontinuities It's annoying.
2015-10-28 22:56:31 +00:00
}
audio: add AO deviation logging Pretty dumb (and doesn't handle pausing or other discontinuities), but at least somewhat idiot-proof.
2015-10-08 21:08:20 +00:00
double delay = ao_get_delay(mpctx->ao);
if (!mpctx->audio_stat_start) {
mpctx->audio_stat_start = mp_time_us();
mpctx->written_audio = delay;
}
double current_audio = mpctx->written_audio - delay;
double current_time = (mp_time_us() - mpctx->audio_stat_start) / 1e6;
MP_STATS(mpctx, "value %f ao-dev", current_audio - current_time);
}
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
// Return the number of samples that must be skipped or prepended to reach the
// target audio pts after a seek (for A/V sync or hr-seek).
// Return value (*skip):
// >0: skip this many samples
// =0: don't do anything
// <0: prepend this many samples of silence
// Returns false if PTS is not known yet.
static bool get_sync_samples(struct MPContext *mpctx, int *skip)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
{
struct MPOpts *opts = mpctx->opts;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
*skip = 0;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
if (mpctx->audio_status != STATUS_SYNCING)
return true;
audio: move decoder context from sh_audio into new struct Move all state that basically changes during decoding or is needed in order to manage decoding itself into a new struct (dec_audio). sh_audio (defined in stheader.h) is supposed to be the audio stream header. This should reflect the file headers for the stream. Putting the decoder context there is strange design, to say the least.
2013-11-23 20:22:17 +00:00
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
struct mp_audio out_format = {0};
ao_get_format(mpctx->ao, &out_format);
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
double play_samplerate = out_format.rate / mpctx->audio_speed;
audio/out: make ao struct opaque We want to move the AO to its own thread. There's no technical reason for making the ao struct opaque to do this. But it helps us sleep at night, because we can control access to shared state better.
2014-03-07 14:24:32 +00:00
audio: cut audio with spdif too on playback restart When playback is started after seeking or opening a file, we need to make sure audio and video line up exactly. This is done by cutting or padding the audio stream to start on the video PTS. This does not quite work with spdif: audio is compressed data, within a spdif frame. There is no way to cut the audio "in between" the frames. Cutting between the frames would just produce broken spdif packets, and who knows how receivers will react to this (play noise?). But we still can cut it in frame boundaries. Unfortunately, we also insert 0 data for "silence" - we probably shouldn't do this. Chances are the receiver will switch to PCM or so. But for now this will have to do. Note that this could be simplified somewhat, as soon as we work with frames. See previous commit.
2015-03-10 14:17:57 +00:00
if (!opts->initial_audio_sync) {
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
mpctx->audio_status = STATUS_FILLING;
return true;
}
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
double written_pts = written_audio_pts(mpctx);
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
if (written_pts == MP_NOPTS_VALUE &&
!mp_audio_buffer_samples(mpctx->ao_chain->ao_buffer))
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
return false; // no audio read yet
player: use different variable to indicate coverart Slightly better.
2016-02-01 21:14:32 +00:00
bool sync_to_video = mpctx->vo_chain && !mpctx->vo_chain->is_coverart &&
player: enable hr-seek on audio after video end Some files can have audio after video has ended, and playback of the audio-only remainder is supposed to work just fine. Seeking is broken-ish though. Not much can be done about this, since it's the way demuxers work. Also, such files are obscure corner cases. But enabling hr-seek for audio after video end can improve the situation a lot. This helps with issue #1533. The reported also provided a command line to produce such a file: ffmpeg -i image.jpg -i audio.flac -threads $(nproc) \ -c:v libvpx -crf 10 -qmin 5 -qmax 55 \ -vf scale=360:-1 -sws_flags lanczos -c:a libvorbis -ac 2 \ -b:a 128K out.webm
2015-01-28 18:36:46 +00:00
mpctx->video_status != STATUS_EOF;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
double sync_pts = MP_NOPTS_VALUE;
if (sync_to_video) {
player: minor simplification in A/V-sync related code Just minor things.
2015-01-30 22:49:30 +00:00
if (mpctx->video_status < STATUS_READY)
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
return false; // wait until we know a video PTS
player: fix initial audio sync in certain cases Regression caused by commit 3b95dd47. Also see commit 4c25b000. We can either use video_next_pts and add "delay", or we just use video_pts. Any other combination breaks. The reason why the assumption that delay==0 at this point was wrong exactly because after displaying the first video frame (usually done before audio resync) a new frame might be "added" immediately, resulting in a new video_next_pts and "delay", which will still amount to video_pts. Fixes #2770. (The reason why display-sync was blamed in this issue is because enabling display-sync in the options forces a prefetch by 2 instead of 1 frames for seeks/playback restart, which triggers the issue, even if display-sync is not actually enabled. In this case, display-sync is never enabled because the frames have a unusually high frame duration. This is also what exposed the initial desync issue.)
2016-01-29 21:43:59 +00:00
if (mpctx->video_pts != MP_NOPTS_VALUE)
sync_pts = mpctx->video_pts - opts->audio_delay;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
} else if (mpctx->hrseek_active) {
sync_pts = mpctx->hrseek_pts;
player: fix mid-stream audio sync condition This code is for resyncing audio-only streams (e.g. switching between audio tracks if no video track is active). This must not be run if the video PTS just isn't known yet. (Although the case in which this changes anything is probably very obscure, if it can even happen. Still, it's a bit more correct.) This is a correction to commit 91a3bda6.
2016-08-07 14:00:14 +00:00
} else {
// If audio-only is enabled mid-stream during playback, sync accordingly.
player: sync audio as well when enabling it mid-stream If an audio track is enabled during playback, then make it resume at the exact "current position", instead of playing audio before that position. This was already done for video.
2016-08-06 15:41:44 +00:00
sync_pts = mpctx->playback_pts;
player: fix mid-stream audio sync condition This code is for resyncing audio-only streams (e.g. switching between audio tracks if no video track is active). This must not be run if the video PTS just isn't known yet. (Although the case in which this changes anything is probably very obscure, if it can even happen. Still, it's a bit more correct.) This is a correction to commit 91a3bda6.
2016-08-07 14:00:14 +00:00
}
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
if (sync_pts == MP_NOPTS_VALUE) {
mpctx->audio_status = STATUS_FILLING;
return true; // syncing disabled
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
double ptsdiff = written_pts - sync_pts;
// Missing timestamp, or PTS reset, or just broken.
player: detect audio PTS jumps, make video PTS heuristic less aggressive This is another attempt at making files with sparse video frames work better. The problem is that you generally can't know whether a jump in video timestamps is just a (very) long video frame, or a timestamp reset. Due to the existence of files with sparse video frames (new frame only every few seconds or longer), every heuristic will be arbitrary (in general, at least). But we can use the fact that if video is continuous, audio should also be continuous. Audio discontinuities can be easily detected, and if that happens, reset some of the playback state. The way the playback state is reset is rather radical (resets decoders as well), but it's just better not to cause too much obscure stuff to happen here. If the A/V sync code were to be rewritten, it should probably strictly use PTS values (not this strange time_frame/delay stuff), which would make it much easier to detect such situations and to react to them.
2016-01-09 19:27:03 +00:00
if (written_pts == MP_NOPTS_VALUE) {
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
MP_WARN(mpctx, "Failed audio resync.\n");
mpctx->audio_status = STATUS_FILLING;
return true;
}
player: detect audio PTS jumps, make video PTS heuristic less aggressive This is another attempt at making files with sparse video frames work better. The problem is that you generally can't know whether a jump in video timestamps is just a (very) long video frame, or a timestamp reset. Due to the existence of files with sparse video frames (new frame only every few seconds or longer), every heuristic will be arbitrary (in general, at least). But we can use the fact that if video is continuous, audio should also be continuous. Audio discontinuities can be easily detected, and if that happens, reset some of the playback state. The way the playback state is reset is rather radical (resets decoders as well), but it's just better not to cause too much obscure stuff to happen here. If the A/V sync code were to be rewritten, it should probably strictly use PTS values (not this strange time_frame/delay stuff), which would make it much easier to detect such situations and to react to them.
2016-01-09 19:27:03 +00:00
ptsdiff = MPCLAMP(ptsdiff, -3600, 3600);
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
player: gross hack to improve non-hr seeking with external audio tracks Relative seeks backwards with external audio tracks does not always work well: it tends to happen that video seek back further than audio, so audio will remain silent until the audio's after-seek position is reached. This happens because we strictly seek both video and audio demuxer to the approximate desirted target PTS, and then start decoding from that. Commit 81358380 removes an older method that was supposed to deal with this. It was sort of bad, because it could lead to the playback core freezing by waiting on network. Ideally, the demuxer layer would probably somehow deal with such seeks, and do them in a way the audio is seeked after video. Currently this is infeasible, because the demuxer layer assumes a single demuxer, and external tracks simply use separate demuxer layers. (MPlayer actually had a pseudo-demuxer that joined external tracks into a single demuxer, but this is not flexible enough - and also, the demuxer layer as it currently exists can't deal with dynamically removing external tracks either. Maybe some time in the future.) Instead, add a gross hack, that essentially reseeks the audio if it detects that it's too far off. The result is actually not too bad, because we can reuse the mechanism that is used for instant track switching. This way we can make sure of the right position, without having to care about certain other issues. It should be noted that if the audio demuxer is used for other tracks too, and the demuxer does not support refresh seeking, audio will probably be off by even a higher amount. But this should be rare.
2016-08-07 14:29:13 +00:00
// Heuristic: if audio is "too far" ahead, and one of them is a separate
// track, allow a refresh seek to the correct position to fix it.
if (ptsdiff > 0.2 && mpctx->audio_allow_second_chance_seek && sync_to_video) {
struct ao_chain *ao_c = mpctx->ao_chain;
if (ao_c && ao_c->track && mpctx->vo_chain && mpctx->vo_chain->track &&
ao_c->track->demuxer != mpctx->vo_chain->track->demuxer)
{
struct track *track = ao_c->track;
double pts = mpctx->video_pts;
if (pts != MP_NOPTS_VALUE)
pts += get_track_seek_offset(mpctx, track);
// (disable it first to make it take any effect)
demuxer_select_track(track->demuxer, track->stream, pts, false);
demuxer_select_track(track->demuxer, track->stream, pts, true);
reset_audio_state(mpctx);
MP_VERBOSE(mpctx, "retrying audio seek\n");
return false;
}
}
mpctx->audio_allow_second_chance_seek = false;
audio: cut audio with spdif too on playback restart When playback is started after seeking or opening a file, we need to make sure audio and video line up exactly. This is done by cutting or padding the audio stream to start on the video PTS. This does not quite work with spdif: audio is compressed data, within a spdif frame. There is no way to cut the audio "in between" the frames. Cutting between the frames would just produce broken spdif packets, and who knows how receivers will react to this (play noise?). But we still can cut it in frame boundaries. Unfortunately, we also insert 0 data for "silence" - we probably shouldn't do this. Chances are the receiver will switch to PCM or so. But for now this will have to do. Note that this could be simplified somewhat, as soon as we work with frames. See previous commit.
2015-03-10 14:17:57 +00:00
int align = af_format_sample_alignment(out_format.format);
audio: strictly align audio on spdif frames We still have a sample-based buffer between filters and audio outputs. In order to avoid cutting frames into half (which can upset receivers), we strictly need to align the boundaries on which we cut the audio.
2015-11-04 15:47:11 +00:00
*skip = (int)(-ptsdiff * play_samplerate) / align * align;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
return true;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
static bool copy_output(struct MPContext *mpctx, struct mp_audio_buffer *outbuf,
int minsamples, double endpts, bool eof, bool *seteof)
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
{
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
struct af_stream *afs = mpctx->ao_chain->af;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
while (mp_audio_buffer_samples(outbuf) < minsamples) {
if (af_output_frame(afs, eof) < 0)
return true; // error, stop doing stuff
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
int cursamples = mp_audio_buffer_samples(outbuf);
int maxsamples = INT_MAX;
if (endpts != MP_NOPTS_VALUE) {
double rate = afs->output.rate / mpctx->audio_speed;
double curpts = written_audio_pts(mpctx);
if (curpts != MP_NOPTS_VALUE)
maxsamples = (endpts - curpts - mpctx->opts->audio_delay) * rate;
}
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct mp_audio *mpa = af_read_output_frame(afs);
if (!mpa)
return false; // out of data
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
if (cursamples + mpa->samples > maxsamples) {
if (cursamples < maxsamples) {
struct mp_audio pre = *mpa;
pre.samples = maxsamples - cursamples;
mp_audio_buffer_append(outbuf, &pre);
mp_audio_skip_samples(mpa, pre.samples);
}
af_unread_output_frame(afs, mpa);
*seteof = true;
return true;
}
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
mp_audio_buffer_append(outbuf, mpa);
talloc_free(mpa);
}
return true;
}
static int decode_new_frame(struct ao_chain *ao_c)
{
if (ao_c->input_frame)
return AD_OK;
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
int res = DATA_EOF;
if (ao_c->filter_src) {
res = lavfi_request_frame_a(ao_c->filter_src, &ao_c->input_frame);
} else if (ao_c->audio_src) {
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
audio_work(ao_c->audio_src);
res = audio_get_frame(ao_c->audio_src, &ao_c->input_frame);
}
switch (res) {
audio/video: merge decoder return values Will be helpful for the coming filter support. I planned on merging audio/video decoding, but this will have to wait a bit longer, so only remove the duplicate status codes.
2016-02-01 20:32:01 +00:00
case DATA_OK: return AD_OK;
case DATA_WAIT: return AD_WAIT;
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
case DATA_AGAIN: return AD_NO_PROGRESS;
audio/video: merge decoder return values Will be helpful for the coming filter support. I planned on merging audio/video decoding, but this will have to wait a bit longer, so only remove the duplicate status codes.
2016-02-01 20:32:01 +00:00
case DATA_EOF: return AD_EOF;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
default: abort();
}
}
/* Try to get at least minsamples decoded+filtered samples in outbuf
* (total length including possible existing data).
* Return 0 on success, or negative AD_* error code.
* In the former case outbuf has at least minsamples buffered on return.
* In case of EOF/error it might or might not be. */
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
static int filter_audio(struct MPContext *mpctx, struct mp_audio_buffer *outbuf,
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
int minsamples)
{
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
struct ao_chain *ao_c = mpctx->ao_chain;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
struct af_stream *afs = ao_c->af;
if (afs->initialized < 1)
return AD_ERR;
MP_STATS(ao_c, "start audio");
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
double endpts = get_play_end_pts(mpctx);
bool eof = false;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
int res;
while (1) {
res = 0;
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
if (copy_output(mpctx, outbuf, minsamples, endpts, false, &eof))
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
break;
res = decode_new_frame(ao_c);
audio: stop being dumb Obvious mistake: we entered EOF drain mode if the decoder returned AD_WAIT, which is very wrong. AD_WAIT means we should retry after waiting for a while (or to be precise, until the demuxer/decoder have more data). We should just pass down this status, and not change the audio chain state. This was exposed by a libavfilter EOF handling bug. Feeding a filter chain with af_dynaudnorm, and sending an EOF before a frame is returned makes it stuck and keeps returning EAGAIN, instead of returning the buffered audio. In combination with the bug at hand, which entered EOG drain mode, it could happen that it got stuck due to libavfilter discarding buffered data each time the demuxer ran out of data. Fixes #3997.
2017-01-08 11:50:26 +00:00
if (res == AD_NO_PROGRESS || res == AD_WAIT)
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
break;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
if (res < 0) {
// drain filters first (especially for true EOF case)
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
copy_output(mpctx, outbuf, minsamples, endpts, true, &eof);
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
break;
}
// On format change, make sure to drain the filter chain.
if (!mp_audio_config_equals(&afs->input, ao_c->input_frame)) {
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
copy_output(mpctx, outbuf, minsamples, endpts, true, &eof);
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
res = AD_NEW_FMT;
break;
}
struct mp_audio *mpa = ao_c->input_frame;
ao_c->input_frame = NULL;
audio: move pts reset check Reduces the dependency of the filter/output code on the decoder.
2016-01-29 21:44:20 +00:00
if (mpa->pts == MP_NOPTS_VALUE) {
ao_c->pts = MP_NOPTS_VALUE;
} else {
// Attempt to detect jumps in PTS. Even for the lowest sample rates
// and with worst container rounded timestamp, this should be a
// margin more than enough.
audio: force pts_reset only when pts jumps forward more than 5s i've seen several mpegts samples where pts jumps backwards and repeats itself. this usually happens on live tv streams from cable providers, particularly when the stream switches from one advertisement to another.
2016-10-20 18:11:56 +00:00
double desync = mpa->pts - ao_c->pts;
if (ao_c->pts != MP_NOPTS_VALUE && fabs(desync) > 0.1) {
audio: move pts reset check Reduces the dependency of the filter/output code on the decoder.
2016-01-29 21:44:20 +00:00
MP_WARN(ao_c, "Invalid audio PTS: %f -> %f\n",
ao_c->pts, mpa->pts);
audio: use brutal resync only on larger PTS discontinuities Let's fix broken samples with questionable heuristic without real reasoning. Until this gets fixed properly, this is a good compromise, though. A proper fix would properly resync audio and video without brutally resetting the decoders, but on the other hand not doing the brutal reset would cause issues in other obscure corner cases such resyncing might cause.
2016-01-31 21:10:11 +00:00
if (desync >= 5)
ao_c->pts_reset = true;
audio: move pts reset check Reduces the dependency of the filter/output code on the decoder.
2016-01-29 21:44:20 +00:00
}
ao_c->pts = mpa->pts + mpa->samples / (double)mpa->rate;
}
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
if (af_filter_frame(afs, mpa) < 0)
return AD_ERR;
}
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
if (res == 0 && mp_audio_buffer_samples(outbuf) < minsamples && eof)
res = AD_EOF;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
MP_STATS(ao_c, "end audio");
return res;
}
audio: move some fallback handling to common AO reload function Now a reload requested by an AO behaves in exactly the same way as changing an AO-related options (like --audio-channels or --audio-exclusive). This is good for testing and uniform behavior. (You could go as far as saying it's a necessity, because the spotty and obscure AO reload behavior is hard to reproduce and thus hard to test at all.)
2016-10-05 14:42:43 +00:00
void reload_audio_output(struct MPContext *mpctx)
{
if (!mpctx->ao)
return;
ao_reset(mpctx->ao);
uninit_audio_out(mpctx);
reinit_audio_filters(mpctx); // mostly to issue refresh seek
// Whether we can use spdif might have changed. If we failed to use spdif
// in the previous initialization, try it with spdif again (we'll fallback
// to PCM again if necessary).
struct ao_chain *ao_c = mpctx->ao_chain;
if (ao_c) {
struct dec_audio *d_audio = ao_c->audio_src;
if (d_audio && ao_c->spdif_failed) {
ao_c->spdif_passthrough = true;
ao_c->spdif_failed = false;
d_audio->try_spdif = true;
ao_c->af->initialized = 0;
if (!audio_init_best_codec(d_audio)) {
MP_ERR(mpctx, "Error reinitializing audio.\n");
error_on_track(mpctx, ao_c->track);
}
}
}
mp_wakeup_core(mpctx);
}
player: minor simplification No need to pass endpts down in such a dumb way. Also remove an outdated comment somewhere.
2016-02-27 19:02:51 +00:00
void fill_audio_out_buffers(struct MPContext *mpctx)
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
{
struct MPOpts *opts = mpctx->opts;
audio: avoid missed wakeups with ab-loops Could get "stuck".
2016-08-24 10:14:48 +00:00
bool was_eof = mpctx->audio_status == STATUS_EOF;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
audio: add AO deviation logging Pretty dumb (and doesn't handle pausing or other discontinuities), but at least somewhat idiot-proof.
2015-10-08 21:08:20 +00:00
dump_audio_stats(mpctx);
audio: move some fallback handling to common AO reload function Now a reload requested by an AO behaves in exactly the same way as changing an AO-related options (like --audio-channels or --audio-exclusive). This is good for testing and uniform behavior. (You could go as far as saying it's a necessity, because the spotty and obscure AO reload behavior is hard to reproduce and thus hard to test at all.)
2016-10-05 14:42:43 +00:00
if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD))
reload_audio_output(mpctx);
audio/out: make ao_request_reload() idempotent This is what you would expect. Before this commit, each ao_request_reload() call would just queue a reload command, and then recreate the AO for the number of times the function was called. Instead of sending a command, introduce some sort of event retrieval mechanism. At least for the reload case, use atomics, because we're too lazy to setup an extra mutex.
2014-11-09 08:58:44 +00:00
audio: move some fallback handling to common AO reload function Now a reload requested by an AO behaves in exactly the same way as changing an AO-related options (like --audio-channels or --audio-exclusive). This is good for testing and uniform behavior. (You could go as far as saying it's a necessity, because the spotty and obscure AO reload behavior is hard to reproduce and thus hard to test at all.)
2016-10-05 14:42:43 +00:00
struct ao_chain *ao_c = mpctx->ao_chain;
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
if (!ao_c)
player: move video display code out of the playloop Basically move the code from playloop.c to video.c. The new function write_video() now contains the code that was part of run_playloop(). There are no functional changes, except handling "new_frame_shown" slightly differently. This is done so that we don't need new a new MPContext field or a return value for write_video() to signal this condition. Instead, it's handled indirectly.
2014-07-30 21:24:08 +00:00
return;
audio: move decoder context from sh_audio into new struct Move all state that basically changes during decoding or is needed in order to manage decoding itself into a new struct (dec_audio). sh_audio (defined in stheader.h) is supposed to be the audio stream header. This should reflect the file headers for the stream. Putting the decoder context there is strange design, to say the least.
2013-11-23 20:22:17 +00:00
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
if (ao_c->af->initialized < 1 || !mpctx->ao) {
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
// Probe the initial audio format. Returns AD_OK (and does nothing) if
// the format is already known.
audio: refactor: work towards unentangling audio decoding and filtering Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
2016-01-21 21:10:15 +00:00
int r = decode_new_frame(mpctx->ao_chain);
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
if (r == AD_WAIT)
audio: cosmetics: remove unused return value
2014-07-29 22:24:57 +00:00
return; // continue later when new data is available
audio: fix EOF handling if nothing could be decoded at all The code for decoding the initial frame has to handle this explicitly.
2016-02-06 17:33:31 +00:00
if (r == AD_EOF) {
mpctx->audio_status = STATUS_EOF;
return;
}
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
reinit_audio_filters_and_output(mpctx);
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
audio: cosmetics: remove unused return value
2014-07-29 22:24:57 +00:00
return; // try again next iteration
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
player: add --audio-wait-open options Complements the option added in the previous commit.
2016-08-09 14:26:44 +00:00
if (ao_c->ao_resume_time > mp_time_sec()) {
double remaining = ao_c->ao_resume_time - mp_time_sec();
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_set_timeout(mpctx, remaining);
player: add --audio-wait-open options Complements the option added in the previous commit.
2016-08-09 14:26:44 +00:00
return;
}
audio: move pts reset check Reduces the dependency of the filter/output code on the decoder.
2016-01-29 21:44:20 +00:00
if (mpctx->vo_chain && ao_c->pts_reset) {
player: detect audio PTS jumps, make video PTS heuristic less aggressive This is another attempt at making files with sparse video frames work better. The problem is that you generally can't know whether a jump in video timestamps is just a (very) long video frame, or a timestamp reset. Due to the existence of files with sparse video frames (new frame only every few seconds or longer), every heuristic will be arbitrary (in general, at least). But we can use the fact that if video is continuous, audio should also be continuous. Audio discontinuities can be easily detected, and if that happens, reset some of the playback state. The way the playback state is reset is rather radical (resets decoders as well), but it's just better not to cause too much obscure stuff to happen here. If the A/V sync code were to be rewritten, it should probably strictly use PTS values (not this strange time_frame/delay stuff), which would make it much easier to detect such situations and to react to them.
2016-01-09 19:27:03 +00:00
MP_VERBOSE(mpctx, "Reset playback due to audio timestamp reset.\n");
reset_playback_state(mpctx);
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
player: detect audio PTS jumps, make video PTS heuristic less aggressive This is another attempt at making files with sparse video frames work better. The problem is that you generally can't know whether a jump in video timestamps is just a (very) long video frame, or a timestamp reset. Due to the existence of files with sparse video frames (new frame only every few seconds or longer), every heuristic will be arbitrary (in general, at least). But we can use the fact that if video is continuous, audio should also be continuous. Audio discontinuities can be easily detected, and if that happens, reset some of the playback state. The way the playback state is reset is rather radical (resets decoders as well), but it's just better not to cause too much obscure stuff to happen here. If the A/V sync code were to be rewritten, it should probably strictly use PTS values (not this strange time_frame/delay stuff), which would make it much easier to detect such situations and to react to them.
2016-01-09 19:27:03 +00:00
return;
}
audio: add a mode to insert silence on severe A/V desync This is probably a stupid idea, but it can't be denied that this actually allows playing video without larger desync, even if video is too slow.
2014-08-15 21:52:42 +00:00
struct mp_audio out_format = {0};
ao_get_format(mpctx->ao, &out_format);
player: separate controls for user and video controlled speed For video sync, we want separate playback speed controls for user- requested speed and the "correction" speed for video timing. Further, we use this separation to make sure only a resampler is inserted if playback speed is only changed for video sync correction. As of this commit, this is basically inactive code. It's just preparation for the video sync code (the following commit).
2015-08-10 16:40:16 +00:00
double play_samplerate = out_format.rate / mpctx->audio_speed;
audio: strictly align audio on spdif frames We still have a sample-based buffer between filters and audio outputs. In order to avoid cutting frames into half (which can upset receivers), we strictly need to align the boundaries on which we cut the audio.
2015-11-04 15:47:11 +00:00
int align = af_format_sample_alignment(out_format.format);
audio: add a mode to insert silence on severe A/V desync This is probably a stupid idea, but it can't be denied that this actually allows playing video without larger desync, even if video is too slow.
2014-08-15 21:52:42 +00:00
audio: fix A/V sync in encoding mode In encoding mode, the AO pretends to be infinitely fast (it will take whatever we write, without ever rejecting input). Commit 261506e3 broke this somehow. It turns out an old hack dealing with this was accidentally dropped. This is the hunk of code whose semantics were (partially) dropped: if (mpctx->d_audio && (mpctx->restart_playback ? !video_left : ao_untimed(mpctx->ao) && (mpctx->delay <= 0 || !video_left))) { int status = fill_audio_out_buffers(mpctx, endpts); // Not at audio stream EOF yet audio_left = status > -2; } This if condition is pretty wild, and it looked like it was pretty much for audio-only mode, rather than subtle handling for encoding mode.
2014-07-31 02:49:24 +00:00
// If audio is infinitely fast, somehow try keeping approximate A/V sync.
if (mpctx->audio_status == STATUS_PLAYING && ao_untimed(mpctx->ao) &&
audio: simplify condition The expression added with the previous commit (0cce8fe6) looked slightly more complicated than it has to be. The code is equivalent.
2014-07-31 19:11:49 +00:00
mpctx->video_status != STATUS_EOF && mpctx->delay > 0)
audio: fix A/V sync in encoding mode In encoding mode, the AO pretends to be infinitely fast (it will take whatever we write, without ever rejecting input). Commit 261506e3 broke this somehow. It turns out an old hack dealing with this was accidentally dropped. This is the hunk of code whose semantics were (partially) dropped: if (mpctx->d_audio && (mpctx->restart_playback ? !video_left : ao_untimed(mpctx->ao) && (mpctx->delay <= 0 || !video_left))) { int status = fill_audio_out_buffers(mpctx, endpts); // Not at audio stream EOF yet audio_left = status > -2; } This if condition is pretty wild, and it looked like it was pretty much for audio-only mode, rather than subtle handling for encoding mode.
2014-07-31 02:49:24 +00:00
return;
player: don't treat audio playback restart while paused special I guess this was supposed to be some sort of optimization, but even though it probably works, it's pretty meaningless and I couldn't measure a difference. One special case killed.
2015-01-30 22:54:43 +00:00
int playsize = ao_get_space(mpctx->ao);
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
int skip = 0;
bool sync_known = get_sync_samples(mpctx, &skip);
if (skip > 0) {
playsize = MPMIN(skip + 1, MPMAX(playsize, 2500)); // buffer extra data
} else if (skip < 0) {
playsize = MPMAX(1, playsize + skip); // silence will be prepended
}
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: add audio drop/duplicate mode Not very robust in the moment.
2015-10-27 19:56:46 +00:00
int skip_duplicate = 0; // >0: skip, <0: duplicate
player: limit speed change in display-sync adrop mode Discontinuities (like toggling fullscreen) can cause multiple frames to be dropped in succession, which sounds very weird. It's better to drop some video frames instead to compensate for larger desyncs. We roughly base it on the maximum allowed speed changes (audio change is "additional" to the video change to account for deviations when playing at max. video speed change).
2015-11-03 19:29:25 +00:00
double drop_limit =
(opts->sync_max_audio_change + opts->sync_max_video_change) / 100;
player: add audio drop/duplicate mode Not very robust in the moment.
2015-10-27 19:56:46 +00:00
if (mpctx->display_sync_active && opts->video_sync == VS_DISP_ADROP &&
fabs(mpctx->last_av_difference) >= opts->sync_audio_drop_size &&
player: limit speed change in display-sync adrop mode Discontinuities (like toggling fullscreen) can cause multiple frames to be dropped in succession, which sounds very weird. It's better to drop some video frames instead to compensate for larger desyncs. We roughly base it on the maximum allowed speed changes (audio change is "additional" to the video change to account for deviations when playing at max. video speed change).
2015-11-03 19:29:25 +00:00
mpctx->audio_drop_throttle < drop_limit &&
player: add audio drop/duplicate mode Not very robust in the moment.
2015-10-27 19:56:46 +00:00
mpctx->audio_status == STATUS_PLAYING)
{
int samples = ceil(opts->sync_audio_drop_size * play_samplerate);
samples = (samples + align / 2) / align * align;
skip_duplicate = mpctx->last_av_difference >= 0 ? -samples : samples;
player: simplify display-adrop mode safeguard It's not needed, because the additional data is not appended, but is the total size of the audio buffer. The maximum size is the static audio drop size (or twice, if the audio is duplicated).
2015-10-28 22:50:59 +00:00
playsize = MPMAX(playsize, samples);
player: limit speed change in display-sync adrop mode Discontinuities (like toggling fullscreen) can cause multiple frames to be dropped in succession, which sounds very weird. It's better to drop some video frames instead to compensate for larger desyncs. We roughly base it on the maximum allowed speed changes (audio change is "additional" to the video change to account for deviations when playing at max. video speed change).
2015-11-03 19:29:25 +00:00
mpctx->audio_drop_throttle += 1 - drop_limit - samples / play_samplerate;
player: add audio drop/duplicate mode Not very robust in the moment.
2015-10-27 19:56:46 +00:00
}
audio: strictly align audio on spdif frames We still have a sample-based buffer between filters and audio outputs. In order to avoid cutting frames into half (which can upset receivers), we strictly need to align the boundaries on which we cut the audio.
2015-11-04 15:47:11 +00:00
playsize = playsize / align * align;
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
int status = mpctx->audio_status >= STATUS_DRAINING ? AD_EOF : AD_OK;
audio: fix resync issue different Commit 10915000 attempted to fix wasting CPU when resyncing and no new data was actually coming from the demuxer. The fix assumed that at this point it would have reached the sync point, but since the code attempts weird incremental decoding, this wasn't actually true. So it broke seeking in addition to removing the CPU waste. Try something else. This time, we essentially only wakeup again if data was read (i.e. audio_decode() returned successfully).
2015-05-19 21:23:17 +00:00
bool working = false;
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
if (playsize > mp_audio_buffer_samples(ao_c->ao_buffer)) {
audio: improve aspects of EOF handling The code actually kept going out of EOF mode into resync mode back into EOF mode when the playloop had to wait after an audio EOF caused by the endpts. This would break seamless looping (as added by the next commit). Apply endpts earlier, to ensure the filter_audio() function always returns AD_EOF in this case. The idiotic ao_buffer makes this an amazing pain in the ass.
2016-08-18 18:38:09 +00:00
status = filter_audio(mpctx, ao_c->ao_buffer, playsize);
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
if (status == AD_WAIT)
audio: cosmetics: remove unused return value
2014-07-29 22:24:57 +00:00
return;
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
if (status == AD_NO_PROGRESS) {
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
player: add complex filter graph support See --lavfi-complex option. This is still quite rough. There's no support for dynamic configuration of any kind. There are probably corner cases where playback might freeze or burn 100% CPU (due to dataflow problems when interaction with libavfilter). Future possible plans might include: - freely switch tracks by providing some sort of default track graph label - automatically enabling audio visualization - automatically mix audio or stack video when multiple tracks are selected at once (similar to how multiple sub tracks can be selected)
2016-02-05 22:19:56 +00:00
return;
}
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
if (status == AD_NEW_FMT) {
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
/* The format change isn't handled too gracefully. A more precise
* implementation would require draining buffered old-format audio
* while displaying video, then doing the output format switch.
*/
audio: add a "weak" gapless mode, and make it default Basically, this allows gapless playback with similar files (including the ordered chapter case), while still being robust in general. The implementation is quite simplistic on purpose, in order to avoid all the weird corner cases that can occur when creating the filter chain. The consequence is that it might do not-gapless playback in more cases when needed, but if that bothers you, you still can use the normal gapless mode. Just using "--gapless-audio" or "--gapless-audio=yes" selects the old mode.
2014-06-08 21:54:05 +00:00
if (mpctx->opts->gapless_audio < 1)
player: remove central uninit_player() function and flags mess Each subsystem (or similar thing) had an INITIALIZED_ flag assigned. The main use of this was that you could pass a bitmask of these flags to uninit_player(). Except in some situations where you wanted to uninitialize nearly everything, this wasn't really useful. Moreover, it was quite annoying that subsystems had most of the code in a specific file, but the uninit code in loadfile.c (because that's where uninit_player() was implemented). Simplify all this. Remove the flags; e.g. instead of testing for the INITIALIZED_AO flag, test whether mpctx->ao is set. Move uninit code to separate functions, e.g. uninit_audio_out().
2014-10-03 17:57:49 +00:00
uninit_audio_out(mpctx);
audio: refactor: separate audio init and filter/output init Before this commit, reinit_audio_chain() did 2 things: create all the management data structures and initialize the decoder, and handling lazy filter/output init (as well as dealing with format changes). For the second purpose, it could be called multiple times (even though it wasn't really idempotent). This was pretty weird, so make them separate functions. The new function is actually idempotent too. It also turns out the reinit functions don't have to call themselves recursively for the spdif PCM fallback.
2016-01-29 21:44:35 +00:00
reinit_audio_filters_and_output(mpctx);
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
audio: cosmetics: remove unused return value
2014-07-29 22:24:57 +00:00
return; // retry on next iteration
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
}
player: don't fall asleep on audio decoding errors This makes it retry later. Fixes #1474.
2015-01-15 20:57:09 +00:00
if (status == AD_ERR)
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
audio: fix resync issue different Commit 10915000 attempted to fix wasting CPU when resyncing and no new data was actually coming from the demuxer. The fix assumed that at this point it would have reached the sync point, but since the code attempts weird incremental decoding, this wasn't actually true. So it broke seeking in addition to removing the CPU waste. Try something else. This time, we essentially only wakeup again if data was read (i.e. audio_decode() returned successfully).
2015-05-19 21:23:17 +00:00
working = true;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
}
audio: better initial sync for files where audio starts later Some files have the first audio much later into the video (for whatever reasons). Instead of appending large amounts of silence to the audio buffer (and refusing to sync if the audio to append is "too large"), just wait until enough video has played.
2014-07-29 22:40:45 +00:00
// If EOF was reached before, but now something can be decoded, try to
// restart audio properly. This helps with video files where audio starts
// later. Retrying is needed to get the correct sync PTS.
audio: fix late audio start Regression since commit bbcd0b6a. This code is just cursed, because it's a fragile state machine with no proper tests, and which could be done in a much simpler way. Without doubt this change will cause a regression in some ridiculous corner case as well. Fixes #3610 (the cause of it, not the behavior it resulted in).
2016-10-02 11:47:19 +00:00
if (mpctx->audio_status >= STATUS_DRAINING &&
mp_audio_buffer_samples(ao_c->ao_buffer) > 0)
{
audio: alternative fix for previous commit This is a somewhat obscure situation, and happens only if audio starts again after it has ended (in particular can happens with files where audio starts later). It doesn't matter much whether audio starts immediately or some milliseconds later, so simplify it.
2014-11-27 17:52:22 +00:00
mpctx->audio_status = STATUS_SYNCING;
audio: better initial sync for files where audio starts later Some files have the first audio much later into the video (for whatever reasons). Instead of appending large amounts of silence to the audio buffer (and refusing to sync if the audio to append is "too large"), just wait until enough video has played.
2014-07-29 22:40:45 +00:00
return; // retry on next iteration
}
audio: better sync behavior on bogus EOF In situations when the demuxer reports EOF, but immediately "recovers" after that and returns new data, it could happen that audio sync was skipped. Deal with this by actually entering the EOF state, instead of assuming this will happen later.
2014-07-30 20:29:24 +00:00
bool end_sync = false;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
if (skip >= 0) {
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
int max = mp_audio_buffer_samples(ao_c->ao_buffer);
mp_audio_buffer_skip(ao_c->ao_buffer, MPMIN(skip, max));
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
// If something is left, we definitely reached the target time.
end_sync |= sync_known && skip < max;
audio: fix a case of going to sleep before playback start This code is tricky because it has to wakeup the mainloop to make progressing during syncing audio, but also has to avoid waking it up when it's not needed. Failure to do so either burns CPU by not ever going to sleep, or causes apparent "freezes" by going to sleep (and it will continue if the mainloop is woken up e.g. due to user input). In this case, simply starting A/V playback with --start=5 and removing an unrelated wakeup in osd.c can trigger such a "freeze". The unrelated wakeup did hide this bug, nonetheless it's a bug. (Can't wait to rewrite this shitty audio resync code. And it's all my fault.)
2016-01-31 21:02:02 +00:00
working |= skip > 0;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
} else if (skip < 0) {
audio: better initial sync for files where audio starts later Some files have the first audio much later into the video (for whatever reasons). Instead of appending large amounts of silence to the audio buffer (and refusing to sync if the audio to append is "too large"), just wait until enough video has played.
2014-07-29 22:40:45 +00:00
if (-skip > playsize) { // heuristic against making the buffer too large
ao_reset(mpctx->ao); // some AOs repeat data on underflow
audio: go to draining state instead of EOF if audio starts later Probably no observable effect, but it's more correct. Setting audio to EOF could have bad effects otherwise (anywhere the player logic for example decides whether EOF was reached, and such).
2014-08-31 12:47:05 +00:00
mpctx->audio_status = STATUS_DRAINING;
audio: better initial sync for files where audio starts later Some files have the first audio much later into the video (for whatever reasons). Instead of appending large amounts of silence to the audio buffer (and refusing to sync if the audio to append is "too large"), just wait until enough video has played.
2014-07-29 22:40:45 +00:00
mpctx->delay = 0;
return;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
}
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
mp_audio_buffer_prepend_silence(ao_c->ao_buffer, -skip);
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
end_sync = true;
}
player: add audio drop/duplicate mode Not very robust in the moment.
2015-10-27 19:56:46 +00:00
if (skip_duplicate) {
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
int max = mp_audio_buffer_samples(ao_c->ao_buffer);
player: add audio drop/duplicate mode Not very robust in the moment.
2015-10-27 19:56:46 +00:00
if (abs(skip_duplicate) > max)
skip_duplicate = skip_duplicate >= 0 ? max : -max;
mpctx->last_av_difference += skip_duplicate / play_samplerate;
if (skip_duplicate >= 0) {
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
mp_audio_buffer_skip(ao_c->ao_buffer, skip_duplicate);
player: add audio drop/duplicate mode Not very robust in the moment.
2015-10-27 19:56:46 +00:00
MP_STATS(mpctx, "drop-audio");
} else {
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
mp_audio_buffer_duplicate(ao_c->ao_buffer, -skip_duplicate);
player: add audio drop/duplicate mode Not very robust in the moment.
2015-10-27 19:56:46 +00:00
MP_STATS(mpctx, "duplicate-audio");
}
MP_VERBOSE(mpctx, "audio skip_duplicate=%d\n", skip_duplicate);
}
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
if (mpctx->audio_status == STATUS_SYNCING) {
if (end_sync)
mpctx->audio_status = STATUS_FILLING;
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
if (status != AD_OK && !mp_audio_buffer_samples(ao_c->ao_buffer))
audio: better sync behavior on bogus EOF In situations when the demuxer reports EOF, but immediately "recovers" after that and returns new data, it could happen that audio sync was skipped. Deal with this by actually entering the EOF state, instead of assuming this will happen later.
2014-07-30 20:29:24 +00:00
mpctx->audio_status = STATUS_EOF;
audio: don't sleep when finishing audio resync This should avoid unnecessary sleeping when audio playback start resync has finished and goes into the normal playback state. This is tricky; see e.g. commit 402fe381.
2015-08-23 19:41:09 +00:00
if (working || end_sync)
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
audio: cosmetics: remove unused return value
2014-07-29 22:24:57 +00:00
return; // continue on next iteration
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
}
assert(mpctx->audio_status >= STATUS_FILLING);
player: really start audio only once video is ready The check whether video is ready yet was done only in STATUS_FILLING. But it also switched to STATUS_READY, which means the next time fill_audio_out_buffers() was called, audio would actually be started before video. In most situations, this bug didn't show up, because it was only triggered if the demuxer didn't provide video packets quickly enough, but did for audio packets. Also log when audio is started. (I hate fill_audio_out_buffers(), why did I write it?)
2016-04-20 11:05:29 +00:00
// We already have as much data as the audio device wants, and can start
// writing it any time.
if (mpctx->audio_status == STATUS_FILLING)
mpctx->audio_status = STATUS_READY;
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
// Even if we're done decoding and syncing, let video start first - this is
// required, because sending audio to the AO already starts playback.
player: really start audio only once video is ready The check whether video is ready yet was done only in STATUS_FILLING. But it also switched to STATUS_READY, which means the next time fill_audio_out_buffers() was called, audio would actually be started before video. In most situations, this bug didn't show up, because it was only triggered if the demuxer didn't provide video packets quickly enough, but did for audio packets. Also log when audio is started. (I hate fill_audio_out_buffers(), why did I write it?)
2016-04-20 11:05:29 +00:00
if (mpctx->audio_status == STATUS_READY) {
if (mpctx->vo_chain && !mpctx->vo_chain->is_coverart &&
mpctx->video_status <= STATUS_READY)
return;
MP_VERBOSE(mpctx, "starting audio playback\n");
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
bool audio_eof = status == AD_EOF;
bool partial_fill = false;
int playflags = 0;
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
if (playsize > mp_audio_buffer_samples(ao_c->ao_buffer)) {
playsize = mp_audio_buffer_samples(ao_c->ao_buffer);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
partial_fill = true;
}
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
audio_eof &= partial_fill;
audio: fix inverted condition Recent regression. Could perhaps make gapless audio fail to work correctly.
2014-08-06 17:25:23 +00:00
// With gapless audio, delay this to ao_uninit. There must be only
// 1 final chunk, and that is handled when calling ao_uninit().
if (audio_eof && !opts->gapless_audio)
playflags |= AOPLAY_FINAL_CHUNK;
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
audio: switch output to mp_audio_buffer Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
2013-11-10 22:38:18 +00:00
struct mp_audio data;
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
mp_audio_buffer_peek(ao_c->ao_buffer, &data);
audio: strictly align audio on spdif frames We still have a sample-based buffer between filters and audio outputs. In order to avoid cutting frames into half (which can upset receivers), we strictly need to align the boundaries on which we cut the audio.
2015-11-04 15:47:11 +00:00
if (audio_eof || data.samples >= align)
data.samples = data.samples / align * align;
audio: fix --end handling (again) Commit c5818046 fixed one case of audio EOF handling, and caused a new one. This time, the ao_buffer doesn't actually contain everyting that should be played - because if --end is used, only a part of it is played. Of course this is stupid, and it will be changed later. For now, this smaller change fixes the bug. Fixes #2189.
2015-08-03 15:02:06 +00:00
data.samples = MPMIN(data.samples, mpctx->paused ? 0 : playsize);
audio: remove an unused parameter
2015-07-24 15:27:40 +00:00
int played = write_to_ao(mpctx, &data, playflags);
player: fix an assert when reinitializing audio in some cases This sometimes happened when changing playback speed (= reinitializing audio) after seeking of playback start. The assertion in audio.c:441 was triggered, because buffer_playable_samples wasn't reset correctly when the audio buffer was cleared or shortened. The assertion is correct and should hold up any time.
2014-02-09 15:21:17 +00:00
assert(played >= 0 && played <= data.samples);
audio: refactor: move MPContext.ao_buffer field It doesn't need to be part of the big context, but is strictly part of shuffling data from the audio filters to audio output, and thus belongs into ao_chain. It also turns out that clearing it in clear_audio_output_buffers() is completely redundant. (Of course ao_buffer is an abomination in the first place and shouldn't exist at all.)
2016-01-21 21:41:41 +00:00
mp_audio_buffer_skip(ao_c->ao_buffer, played);
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
player: fix display-sync adrop speed limiting Commit 49d94853 worked only at the start of playback.
2015-11-04 09:57:07 +00:00
mpctx->audio_drop_throttle =
MPMAX(0, mpctx->audio_drop_throttle - played / play_samplerate);
player: limit speed change in display-sync adrop mode Discontinuities (like toggling fullscreen) can cause multiple frames to be dropped in succession, which sounds very weird. It's better to drop some video frames instead to compensate for larger desyncs. We roughly base it on the maximum allowed speed changes (audio change is "additional" to the video change to account for deviations when playing at max. video speed change).
2015-11-03 19:29:25 +00:00
audio: add AO deviation logging Pretty dumb (and doesn't handle pausing or other discontinuities), but at least somewhat idiot-proof.
2015-10-08 21:08:20 +00:00
dump_audio_stats(mpctx);
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
mpctx->audio_status = STATUS_PLAYING;
audio: fix --end handling (again) Commit c5818046 fixed one case of audio EOF handling, and caused a new one. This time, the ao_buffer doesn't actually contain everyting that should be played - because if --end is used, only a part of it is played. Of course this is stupid, and it will be changed later. For now, this smaller change fixes the bug. Fixes #2189.
2015-08-03 15:02:06 +00:00
if (audio_eof && !playsize) {
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
mpctx->audio_status = STATUS_DRAINING;
// Wait until the AO has played all queued data. In the gapless case,
// we trigger EOF immediately, and let it play asynchronously.
audio: avoid missed wakeups with ab-loops Could get "stuck".
2016-08-24 10:14:48 +00:00
if (ao_eof_reached(mpctx->ao) || opts->gapless_audio) {
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
mpctx->audio_status = STATUS_EOF;
player: specifically log audio EOF too
2017-03-14 14:55:42 +00:00
if (!was_eof) {
MP_VERBOSE(mpctx, "audio EOF reached\n");
player: litter code with explicit wakeup calls This does 3 kinds of changes: - change sleeptime=x to mp_set_timeout() - change sleeptime=0 to mp_wakeup_core() calls (to be more explicit) - change commands etc. to call mp_wakeup_core() if they do changes that require the playloop to be rerun This is preparation for the following changes. The goal is to process client API requests without having to rerun the playloop every time. As of this commit, the changes should not change behavior. In particular, the playloop is still implicitly woken up on every command.
2016-09-16 12:24:15 +00:00
mp_wakeup_core(mpctx);
player: specifically log audio EOF too
2017-03-14 14:55:42 +00:00
}
audio: avoid missed wakeups with ab-loops Could get "stuck".
2016-08-24 10:14:48 +00:00
}
audio: change playback restart and resyncing This commit makes audio decoding non-blocking. If e.g. the network is too slow the playloop will just go to sleep, instead of blocking until enough data is available. For video, this was already done with commit 7083f88c. For audio, it's unfortunately much more complicated, because the audio decoder was used in a blocking manner. Large changes are required to get around this. The whole playback restart mechanism must be turned into a statemachine, especially since it has close interactions with video restart. Lots of video code is thus also changed. (For the record, I don't think switching this code to threads would make this conceptually easier: the code would still have to deal with external input while blocked, so these in-between states do get visible [and thus need to be handled] anyway. On the other hand, it certainly should be possible to modularize this code a bit better.) This will probably cause a bunch of regressions.
2014-07-28 18:40:43 +00:00
}
Split mplayer.c mplayer.c was a bit too big. Split it into multiple files. I hope the way it's split makes sense. Maybe some things don't make too much sense, or go against intuition. These will fixed as soon as I notice them. Some files are a bit questionable (misc.c, osd.c, configfiles.c), and suggestions how to organize this better are welcome. Regressions are possible due to reorganized include statements. Obviously I didn't just copy mplayer.c's orgy of include statements, but recreated them for each file. It's easily possible that there are oversights and mistakes, which will show up on other platforms. There is one actual change: the public avutil.h include is removed from encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I consider using libavutil too much as dangerous, because the set of include files they recursively pull in is rather arbitrary and is different between FFmpeg and Libav.
2013-10-29 21:38:29 +00:00
}
player: factor audio buffer clearing code Note that the change in seek_reset is not entirely equivalent: we even drop the remainder of buffered audio when seeking. This should be more correct, because the whole point of the reset_ao parameter is to control whether audio queued for output should be dropped or not.
2013-11-08 19:02:09 +00:00
// Drop data queued for output, or which the AO is currently outputting.
void clear_audio_output_buffers(struct MPContext *mpctx)
{
player: fix crash on early audio uninit Could crash when exiting playback in very early stages of initialization. CC: @mpv-player/stable
2014-10-15 23:03:02 +00:00
if (mpctx->ao)
player: factor audio buffer clearing code Note that the change in seek_reset is not entirely equivalent: we even drop the remainder of buffered audio when seeking. This should be more correct, because the whole point of the reset_ao parameter is to control whether audio queued for output should be dropped or not.
2013-11-08 19:02:09 +00:00
ao_reset(mpctx->ao);
}
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