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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"
#include "talloc.h"
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
#include "audio/mixer.h"
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: 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: 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
struct af_stream *afs = mpctx->d_audio->afilter;
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)
{
struct af_stream *afs = mpctx->d_audio->afilter;
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
}
static int recreate_audio_filters(struct MPContext *mpctx)
{
assert(mpctx->d_audio);
struct af_stream *afs = mpctx->d_audio->afilter;
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 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
mixer_reinit_audio(mpctx->mixer, mpctx->ao, afs);
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: 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
struct dec_audio *d_audio = mpctx->d_audio;
if (!d_audio)
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
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
af_uninit(mpctx->d_audio->afilter);
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 recreate_audio_filters(mpctx) < 0 ? -1 : 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
player: silence sporadic error messages on audio init When the audio format is not known yet and the audio chain is still initializing, filter reinit will fail. Normally, attempts to reinitialize filters at this stage should be rare (e.g. user commands editing the filter chain). But it sometimes happened with track switching in combination with the video code calling update_playback_speed() at arbitrary times. Get rid of the message by not trying to change the filters for the sake of playback speed update while decoding is still being initialized.
2015-11-10 16:53:05 +00:00
if (!mpctx->d_audio || mpctx->d_audio->afilter->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: 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)
{
if (mpctx->d_audio)
audio_reset_decoding(mpctx->d_audio);
audio: fix obscure audio resync failure with timelines Somehow, there was a larger misunderstanding in the code: ao_buffer does not need to be preserved over audio reinit for proper support of gapless audio. The actual AO internal buffer takes care of this. In fact, preserving ao_buffer just breaks audio resync. In the ordered chapter case, end_pts is used, which means not all audio data in the buffer is played, thus some data is left over when audio decoding resumes on the next segment. This triggers some code that aborts resync if there's "audio decoded" (ao_buffer contains something), but no PTS is known (nothing was actually decoded yet). Simplify, and always bind the output buffer to the decoder. CC: @mpv-player/stable (maybe)
2014-09-04 21:35:11 +00:00
if (mpctx->ao_buffer)
mp_audio_buffer_clear(mpctx->ao_buffer);
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
mpctx->audio_status = mpctx->d_audio ? 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: 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);
audio: fix some issues when reloading the AO We absolutely need to clear the AO reference in the mixer. The audio_status must be changed to a state where no code assumes that the AO is available. (It's allowed to do this blindly.)
2014-11-12 13:15:49 +00:00
mixer_uninit_audio(mpctx->mixer);
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
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;
}
void uninit_audio_chain(struct MPContext *mpctx)
{
if (mpctx->d_audio) {
mixer_uninit_audio(mpctx->mixer);
audio_uninit(mpctx->d_audio);
mpctx->d_audio = NULL;
talloc_free(mpctx->ao_buffer);
mpctx->ao_buffer = NULL;
mpctx->audio_status = STATUS_EOF;
reselect_demux_streams(mpctx);
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
}
}
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
void reinit_audio_chain(struct MPContext *mpctx)
{
struct MPOpts *opts = mpctx->opts;
player: add infrastructure to select multiple tracks at once Of course this does not allow decoding multiple tracks at once; it just adds some minor infrastructure, which could be used to achieve this.
2013-12-24 16:46:08 +00:00
struct track *track = mpctx->current_track[0][STREAM_AUDIO];
player: fix desync when seeking and switching external tracks If you for example use --audio-file, disable the external track, seek, and enable the external track again, the playback position of the external file was off, and you would get major A/V desync. This was actually supposed to work, but broke at some time ago (probably commit 2b87415f). It didn't work, because it attempted to seek the stream if it was already selected, which was always true due to reselect_demux_streams() being called before that. Fix by putting the initial selection and the seek together.
2014-07-29 15:55:28 +00:00
struct sh_stream *sh = track ? track->stream : NULL;
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
if (!sh) {
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);
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 no_audio;
}
client API: add events for video and audio reconfig
2014-02-17 01:52:26 +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
if (!mpctx->d_audio) {
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
mpctx->d_audio = talloc_zero(NULL, struct dec_audio);
audio: mp_msg conversions
2013-12-21 17:23:59 +00:00
mpctx->d_audio->log = mp_log_new(mpctx->d_audio, mpctx->log, "!ad");
mpctx->d_audio->global = mpctx->global;
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
mpctx->d_audio->opts = opts;
mpctx->d_audio->header = sh;
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
mpctx->d_audio->pool = mp_audio_pool_create(mpctx->d_audio);
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
mpctx->d_audio->afilter = af_new(mpctx->global);
mpctx->d_audio->afilter->replaygain_data = sh->audio->replaygain_data;
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
mpctx->d_audio->spdif_passthrough = true;
audio: fix obscure audio resync failure with timelines Somehow, there was a larger misunderstanding in the code: ao_buffer does not need to be preserved over audio reinit for proper support of gapless audio. The actual AO internal buffer takes care of this. In fact, preserving ao_buffer just breaks audio resync. In the ordered chapter case, end_pts is used, which means not all audio data in the buffer is played, thus some data is left over when audio decoding resumes on the next segment. This triggers some code that aborts resync if there's "audio decoded" (ao_buffer contains something), but no PTS is known (nothing was actually decoded yet). Simplify, and always bind the output buffer to the decoder. CC: @mpv-player/stable (maybe)
2014-09-04 21:35:11 +00:00
mpctx->ao_buffer = mp_audio_buffer_create(NULL);
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 (!audio_init_best_codec(mpctx->d_audio))
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
goto init_error;
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
reset_audio_state(mpctx);
audio: correctly initialize output buffer Just like the previous commit, this takes care of fallout from commit 7ab228, which fixed a bug, but introduced some new ones. CC: @mpv-player/stable
2014-09-05 15:48:15 +00:00
if (mpctx->ao) {
struct mp_audio fmt;
ao_get_format(mpctx->ao, &fmt);
mp_audio_buffer_reinit(mpctx->ao_buffer, &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
}
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
assert(mpctx->d_audio);
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: 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
struct mp_audio in_format = mpctx->d_audio->decode_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.
mpctx->sleeptime = 0;
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 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
struct af_stream *afs = mpctx->d_audio->afilter;
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: 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
mp_audio_set_channels(&afs->output, &opts->audio_output_channels);
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: 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) &&
mpctx->d_audio->spdif_passthrough;
bool ao_null_fallback = opts->ao_null_fallback && !spdif_fallback;
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_chmap_remove_useless_channels(&afs->output.channels,
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
&opts->audio_output_channels);
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);
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
mpctx->ao = ao_init_best(mpctx->global, ao_null_fallback, mpctx->input,
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
mpctx->encode_lavc_ctx, afs->output.rate,
afs->output.format, afs->output.channels);
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;
}
}
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.
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
if (spdif_fallback) {
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
mpctx->d_audio->spdif_passthrough = false;
audio: make spdif re-probe from normal decoding work The previous commit handled not falling back to normal decoding if the AO was reloaded (I think...), and this tries to re-engage spdif pass- through if it was previously falling back to normal decoding (e.g. because it temporarily switched to an audio device incapable of passthrough).
2015-10-06 18:17:49 +00:00
mpctx->d_audio->spdif_failed = true;
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 (!audio_init_best_codec(mpctx->d_audio))
goto init_error;
reset_audio_state(mpctx);
reinit_audio_chain(mpctx);
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/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
mp_audio_buffer_reinit(mpctx->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);
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
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;
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: don't print audio/video init failure message twice The messages "Audio: no audio" and "Video: no video" could be printed twice each if initializing them failed. Prevent his silliness. CC: @mpv-player/stable
2014-10-02 01:12:45 +00:00
if (track)
player: fix exiting if both audio and video fail initializing The player was supposed to exit playback if both video and audio failed to initialize (or if one of the streams was not selected when the other stream failed). This didn't work; for one this check was missing from one of the failure paths. And more importantly, both checked the current_track array incorrectly. Fix these issues, and move the failure handling code into a common function. CC: @mpv-player/stable
2014-10-23 16:31:43 +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: 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
struct dec_audio *d_audio = mpctx->d_audio;
if (!d_audio)
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: 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
struct mp_audio in_format = d_audio->decode_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 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 (!mp_audio_config_valid(&in_format) || d_audio->afilter->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: 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
double a_pts = d_audio->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: 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
// d_audio->pts is the timestamp of the latest input packet with
// known pts that the decoder has decoded. d_audio->pts_bytes is
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 amount of bytes the decoder has written after that timestamp.
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
a_pts += d_audio->pts_offset / (double)in_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
// Now a_pts hopefully holds the pts for end of audio from decoder.
// Subtract data in buffers between decoder and audio out.
// Decoded but not filtered
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
if (d_audio->waiting)
a_pts -= d_audio->waiting->samples / (double)in_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
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: 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
double buffered_output = af_calc_delay(d_audio->afilter);
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/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
buffered_output += mp_audio_buffer_seconds(mpctx->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 virtual time for --audio-file with ordered chapters Apparently users prefer this behavior. It was used for subtitles too, so move the code to calculate the video offset into a separate function. Seeking also needs to be fixed. Fixes #1018.
2014-08-15 21:32:37 +00:00
return a_pts +
get_track_video_offset(mpctx, mpctx->current_track[0][STREAM_AUDIO]);
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);
if (written_pts == MP_NOPTS_VALUE && !mp_audio_buffer_samples(mpctx->ao_buffer))
return false; // no audio read yet
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
bool sync_to_video = mpctx->d_video && mpctx->sync_audio_to_video &&
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: minor simplification in A/V-sync related code Just minor things.
2015-01-30 22:49:30 +00:00
if (mpctx->video_next_pts != MP_NOPTS_VALUE)
player: simplify audio sync pts calculation This was done for symmetry with adjust_sync(). But mpctx->delay is always 0 at this point, so prefer slightly simpler code.
2015-10-27 19:56:39 +00:00
sync_pts = mpctx->video_next_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;
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 (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.
audio: increase maximum amount of audio skipped for seeking Precise seeking requires skipping audio, since the demuxer usually doesn't seek precisely enough. There is a sanity check that prevents skipping more than 300 seconds of audio. This still fails with very large mp3s. For example, with a 1GB sized mp3 with Xing headers, entries will be 4 MB apart on average, and occasionally much more. Just bump the limit. I'm not even sure why it was added in the first place; I suppose it's most important for files with real PTS resets.
2015-03-24 14:38:51 +00:00
if (written_pts == MP_NOPTS_VALUE || fabs(ptsdiff) > 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
MP_WARN(mpctx, "Failed audio resync.\n");
mpctx->audio_status = STATUS_FILLING;
return true;
}
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: remove an old hack This was a cosmetic issue. It's handled differently now (clamping the display time to known duration range). This reverts commit 33b57f55573e658b3af6c6e8ff3188c8f959e82e.
2015-05-30 19:58:33 +00:00
void fill_audio_out_buffers(struct MPContext *mpctx, double endpts)
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: 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
struct dec_audio *d_audio = mpctx->d_audio;
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/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
if (mpctx->ao && ao_query_and_reset_events(mpctx->ao, AO_EVENT_RELOAD)) {
ao_reset(mpctx->ao);
uninit_audio_out(mpctx);
audio: re-probe spdif if AO is reloaded Makes the spdif automagic work better on audio hotplugging.
2015-10-06 17:59:02 +00:00
if (d_audio) {
audio: make spdif re-probe from normal decoding work The previous commit handled not falling back to normal decoding if the AO was reloaded (I think...), and this tries to re-engage spdif pass- through if it was previously falling back to normal decoding (e.g. because it temporarily switched to an audio device incapable of passthrough).
2015-10-06 18:17:49 +00:00
if (mpctx->d_audio->spdif_failed) {
mpctx->d_audio->spdif_failed = false;
mpctx->d_audio->spdif_passthrough = true;
if (!audio_init_best_codec(mpctx->d_audio)) {
MP_ERR(mpctx, "Error reinitializing audio.\n");
error_on_track(mpctx, mpctx->current_track[0][STREAM_AUDIO]);
return;
}
}
audio: fix some issues when reloading the AO We absolutely need to clear the AO reference in the mixer. The audio_status must be changed to a state where no code assumes that the AO is available. (It's allowed to do this blindly.)
2014-11-12 13:15:49 +00:00
mpctx->audio_status = STATUS_SYNCING;
audio: re-probe spdif if AO is reloaded Makes the spdif automagic work better on audio hotplugging.
2015-10-06 17:59:02 +00:00
}
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
}
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
if (!d_audio)
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 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 (d_audio->afilter->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.
int r = initial_audio_decode(mpctx->d_audio);
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: handle reinit after AO reload slightly cleaner Don't print bogus messages about packets read in verbose mode.
2014-11-09 08:54:39 +00:00
if (r != AD_OK) {
mpctx->d_audio->init_retries += 1;
if (mpctx->d_audio->init_retries >= 50) {
MP_ERR(mpctx, "Error initializing audio.\n");
player: fix --stop-playback-on-init-failure on audio init failure This was forgotten when the option was implemented, and makes this option work as advertised. Fixes #1473 (though the default behavior is probably still stupid).
2015-01-15 19:12:51 +00:00
error_on_track(mpctx, mpctx->current_track[0][STREAM_AUDIO]);
audio: handle reinit after AO reload slightly cleaner Don't print bogus messages about packets read in verbose mode.
2014-11-09 08:54:39 +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
}
reinit_audio_chain(mpctx);
audio: don't go to sleep after audio reinit It possibly goes to sleep without actually starting to decode audio. Possibly fixes a problem with --no-osc --no-video reported on IRC. CC: @mpv-player/stable
2014-10-16 23:10:43 +00:00
mpctx->sleeptime = 0;
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
}
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: 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 status = 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: 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 (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) {
status = audio_decode(d_audio, mpctx->ao_buffer, playsize);
if (status == AD_WAIT)
audio: cosmetics: remove unused return value
2014-07-29 22:24:57 +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);
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
reinit_audio_chain(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->sleeptime = 0;
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)
mpctx->sleeptime = 0;
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: 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
if (mpctx->audio_status >= STATUS_DRAINING && status == AD_OK) {
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) {
int max = mp_audio_buffer_samples(mpctx->ao_buffer);
mp_audio_buffer_skip(mpctx->ao_buffer, MPMIN(skip, max));
// If something is left, we definitely reached the target time.
end_sync |= sync_known && skip < max;
} 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: 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
mp_audio_buffer_prepend_silence(mpctx->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) {
int max = mp_audio_buffer_samples(mpctx->ao_buffer);
if (abs(skip_duplicate) > max)
skip_duplicate = skip_duplicate >= 0 ? max : -max;
mpctx->last_av_difference += skip_duplicate / play_samplerate;
if (skip_duplicate >= 0) {
mp_audio_buffer_skip(mpctx->ao_buffer, skip_duplicate);
MP_STATS(mpctx, "drop-audio");
} else {
mp_audio_buffer_duplicate(mpctx->ao_buffer, -skip_duplicate);
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: fix initial sync with huge AO buffer With e.g --start=-3 --audio-buffer=10 the decoder entered EOF state before the initial sync was finished, entered STATUS_EOF, and just started playing audio from a random position. This doesn't handle seeking outside of the file, which is a different case. E.g. --start=30:00 with audio and video enabled in a file shorter than 30:00 will play a random last part of audio. This could perhaps be fixed by using the hr-seek target for cutting audio, instead of the video PTS, but that would be kind of intrusive, so don't do it for now. The simpler solution, assuming audio EOF on video EOF, wouldn't work, because we allow audio to start before video, or to last after video.
2014-09-06 11:24:44 +00:00
if (status != AD_OK && !mp_audio_buffer_samples(mpctx->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)
audio: avoid wasting CPU due to continuous wakeup Thsi code path happens during seeking. If video is still being decoded to get to the first video frame, audio has nothing to do, as it is synchronized against the first video frame. We only want to wake up if there's an actual state change. Fixes #1958.
2015-05-19 19:33:48 +00:00
mpctx->sleeptime = 0;
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);
// Even if we're done decoding and syncing, let video start first - this is
// required, because sending audio to the AO already starts playback.
if (mpctx->audio_status == STATUS_FILLING && mpctx->sync_audio_to_video &&
mpctx->video_status <= STATUS_READY)
{
mpctx->audio_status = STATUS_READY;
audio: cosmetics: remove unused return value
2014-07-29 22:24:57 +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
}
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: respect --end/--length with spdif passthrough In theory, we can't really do this, because we don't know when a spdif frame ends. Spdif transports compressed audio through audio setups that were originally designed for PCM only (which includes the audio filter chain, the AO API, most audio output APIs, etc.), and to reach this goal, spdif pretends to be PCM. Compressed data frames are padded with zeros, until a certain data rate is reached, which corresponds to a pseudo-PCM format with 2 bytes per sample and 2 channels at 48000 Hz. Of course an actual spdif frame is significantly larger than a frame of the PCM format it pretends to be, so cutting audio data on frame boundaries (as according to the pseudo-PCM format) merely yields an incomplete and broken frame, not audio that plays for the desired duration. However, sending an incomplete frame might still be much better than the current behavior, which simply ignores --end/--length (but still lets the video end at the exact end time). Should this result in trouble with real spdif receivers, this commit probably has to be reverted.
2013-11-23 20:42:31 +00:00
if (endpts != MP_NOPTS_VALUE) {
player: remove redundant variable mpctx->audio_delay always has the same value as opts->audio_delay. (This was not the case a long time ago, when the audio-delay property didn't actually write to opts->audio_delay. I think.)
2015-01-29 11:10:07 +00:00
double samples = (endpts - written_audio_pts(mpctx) - 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
* play_samplerate;
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
if (playsize > samples) {
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 = MPMAX((int)samples / align * align, 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_eof = true;
partial_fill = true;
}
}
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
if (playsize > mp_audio_buffer_samples(mpctx->ao_buffer)) {
playsize = mp_audio_buffer_samples(mpctx->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/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
mp_audio_buffer_peek(mpctx->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: change handling of an EOF corner case This code handles buggy AOs (even if all AOs are bug-free, it's good for robustness). Move handling of it to the AO feed thread. Now this check doesn't require magic numbers and does exactly what's it supposed to do.
2014-05-30 00:14:38 +00:00
mp_audio_buffer_skip(mpctx->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.
if (ao_eof_reached(mpctx->ao) || opts->gapless_audio)
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
}
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);
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_buffer)
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
mp_audio_buffer_clear(mpctx->ao_buffer);
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
}
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