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* player: use af_scaletempo when slowing down audio tooGravatar wm42015-02-12
| | | | | | In my opinion the artifacts created by af_scaletempo on extreme slowdown (50% or so) are too bothersome - but users disagree. So use af_scaletempo on any speed changes, not just on speedup.
* player: skip audio filter reinit on some types of speed changesGravatar wm42015-02-10
| | | | | | | | | | | | | This avoids potentially dropping some small amount of audio data buffered in filters. Reinit can be skipped only if the filter is af_scaletempo (which maps to AF_CONTROL_SET_PLAYBACK_SPEED). The other case using af_lavrresample is much more complicated due to filter chain politics. Also, changing speed between 1.0 and something higher typically inserts or removes the filter, so this obviously requires reinitialization. It can be prevented by forcing the filter with --af=scaletempo.
* player: don't treat audio playback restart while paused specialGravatar wm42015-01-30
| | | | | | 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.
* player: minor simplification in A/V-sync related codeGravatar wm42015-01-30
| | | | Just minor things.
* player: remove redundant variableGravatar wm42015-01-29
| | | | | | 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.)
* player: enable hr-seek on audio after video endGravatar wm42015-01-28
| | | | | | | | | | | | | | | | | | 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
* audio: don't force any parameters if spdif is usedGravatar wm42015-01-20
| | | | | | The existing code only ignored --audio-channels, but not --audio-rate or --audio-format if spdif passthrough is used. Setting these makes no sense.
* player: don't fall asleep on audio decoding errorsGravatar wm42015-01-15
| | | | | | This makes it retry later. Fixes #1474.
* player: fix --stop-playback-on-init-failure on audio init failureGravatar wm42015-01-15
| | | | | | | 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).
* audio: alternative fix for previous commitGravatar wm42014-11-27
| | | | | | | 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.
* audio: fix busy loop when seeking while pausedGravatar wm42014-11-27
| | | | | | | | | | | | | | | | | | | | When playing paused, the amount of decoded audio is limited to a small amount (1 sample), because we don't write any audio to the AO when paused. The small amount could trigger the case of the wanted audio being too far in the future in the PTS sync code, which set the audio status to STATUS_DRAINING, which in turn triggered the EOF code in the next iteration. This was ok, but unfortunately, this triggered another retry in order to check resuming from EOF by setting the status to STATUS_SYNCING, which in turn lead to the busy loop by alternating between the 2 states. So don't try resyncing while paused. Since the PTS syncing code also calls ao_reset(), this could cause the pulseaudio daemon to consume some CPU time as well. This was caused by commit 33b57f55. Before that, the playloop was merely run more often, but didn't cause any problems. Fixes #1288.
* audio: make mp_audio_config_to_str return a stack-allocated stringGravatar wm42014-11-25
| | | | Simpler overall.
* audio: fix some issues when reloading the AOGravatar wm42014-11-12
| | | | | | | 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.)
* audio: make decoders output refcounted framesGravatar wm42014-11-10
| | | | | | | | | | | | | | 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.
* audio: change how filters are inserted on playback speed changesGravatar wm42014-11-10
| | | | | | | | | | 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().
* audio/out: make ao_request_reload() idempotentGravatar wm42014-11-09
| | | | | | | | | | 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.
* audio: handle reinit after AO reload slightly cleanerGravatar wm42014-11-09
| | | | Don't print bogus messages about packets read in verbose mode.
* player: improve audio time displayGravatar wm42014-11-08
| | | | | | | | | | | | | | | | | | This commit fixes a "cosmetic" user interface issue. Instead of displaying the interpolated seek time on OSD, show the actual audio time. This is rather silly: when seeking in audio-only mode, it takes some iterations until audio is "ready", but on the other hand, the audio state machine is rather fickle, and fixing this cosmetic issue would be intrusive. So just add a hack that paints over the ugly behavior as perceived by the user. Probably the lesser evil. It doesn't happen if video is enabled, because that mode sets the current time immediately to video PTS. (Audio has to be synced to video, so the code is a bit more complex.) Fixes #1233.
* client API: better error reportingGravatar wm42014-10-28
| | | | Give somewhat more information on playback failure.
* player: fix exiting if both audio and video fail initializingGravatar wm42014-10-23
| | | | | | | | | | | | | 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
* audio: don't go to sleep after audio reinitGravatar wm42014-10-17
| | | | | | | 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
* player: fix crash on early audio uninitGravatar wm42014-10-16
| | | | | | | Could crash when exiting playback in very early stages of initialization. CC: @mpv-player/stable
* player: exit if audio init fails and there's no videoGravatar wm42014-10-10
| | | | | | Seems logical. For some reason, the player allows deselecting both audio and video stream without quitting (a deliberate feature of which I have no idea why it was added years ago), so this is needed.
* player: remove central uninit_player() function and flags messGravatar wm42014-10-03
| | | | | | | | | | | | | | 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().
* player: don't print audio/video init failure message twiceGravatar wm42014-10-02
| | | | | | | 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
* audio: enable pitch correction by default when playing fastGravatar wm42014-10-02
| | | | | | | 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.
* command: move setting playback speed to a separate functionGravatar wm42014-10-02
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* audio: refactor some aspects of filter chain setupGravatar wm42014-10-02
| | | | | | | | | | | 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.
* audio: remove --audiodropGravatar wm42014-09-30
| | | | | | | | | | | | | | | This would play some silence in case video was slower than audio. If framedropping is already enabled, there's no other way to keep A/V sync, short of changing audio playback speed (which would give worse results). The --audiodrop option inserted silence if there was more than 500ms desync. This worked somewhat, but I think it was a silly idea after all. Whether the playback experience is really bad or slightly worse doesn't really matter. There also was a subtle bug with PTS handling, that apparently caused A/V desync anyway at ridiculous playback speeds. Just remove this feature; nobody is going to use it anyway.
* audio: cleanup spdif format definitionsGravatar wm42014-09-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Before this commit, there was AF_FORMAT_AC3 (the original spdif format, used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS and DTS-HD), which was handled as some sort of superset for AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used IEC61937-framing, but still was handled as something "separate". Technically, all of them are pretty similar, but may use different bitrates. Since digital passthrough pretends to be PCM (just with special headers that wrap digital packets), this is easily detectable by the higher samplerate or higher number of channels, so I don't know why you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs. AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is just a mess. Simplify this by handling all these formats the same way. AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3). All AOs just accept all spdif formats now - whether that works or not is not really clear (seems inconsistent due to earlier attempts to make DTS-HD work). But on the other hand, enabling spdif requires manual user interaction, so it doesn't matter much if initialization fails in slightly less graceful ways if it can't work at all. At a later point, we will support passthrough with ao_pulse. It seems the PulseAudio API wants to know the codec type (or maybe not - feeding it DTS while telling it it's AC3 works), add separate formats for each codecs. While this reminds of the earlier chaos, it's stricter, and most code just uses AF_FORMAT_IS_IEC61937(). Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to include special formats, so that it always describes the fundamental sample format type. This also ensures valid AF formats are never 0 (this was probably broken in one of the earlier commits from today).
* player: reset last_av_difference if not applicableGravatar wm42014-09-20
| | | | | | Don't let stale values linger around. Also fix a slightly related case in audio.c.
* audio: fix initial sync with huge AO bufferGravatar wm42014-09-06
| | | | | | | | | | | | | | 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.
* audio: correctly initialize output bufferGravatar wm42014-09-05
| | | | | | | 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
* audio: fix obscure audio resync failure with timelinesGravatar wm42014-09-05
| | | | | | | | | | | | | | | | | 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)
* cosmetics: remove a stray ';'Gravatar wm42014-09-05
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* audio: go to draining state instead of EOF if audio starts laterGravatar wm42014-08-31
| | | | | | 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).
* audio: restore old speed change behaviorGravatar wm42014-08-28
| | | | | | | | | | | | | | | | | | | | Don't attempt to resync after speed changes. Note that most other cases of audio reinit (like switching tracks etc.) still resync, but other code paths take care of setting the audio_status accordingly. This restores the old behavior of not trying to fix audio desync, which was probably changed with commit 261506e3. Note that the code as of now wasn't even entirely correct, since the A/V sync values are slightly shifted. The dsync depends on the audio buffer size, so a larger buffer size will show more extreme desync. Also see mplayer2 commit 213a224e, which should fixed this - it was not merged into mpv, because it disabled audio for too long, resulting in a worse user experience. This is similar to the issue this commit attempts to fix. Fixes: #1042 (probably) CC: @mpv-player-stable
* player: minor changesGravatar wm42014-08-25
| | | | | | | | | | | | This shouldn't change anything functionally. Change the A/V desync message. --framedrop is enabled by default now, so the text must be changed a little. I've never heard of audio outputs messing up A/V sync recently, so remove that part. Remove the unused ao_pts field. Reorder 2 A/V sync related expressions so that they look the same.
* audio: minor improvements to timeline switchingGravatar wm42014-08-23
| | | | | | | | In theory, timestamps can be negative, so we shouldn't just return -1 as special value. Remove the separate code for clearing decode buffers; use the same code that is used for normal seek reset.
* player: fix recent speed change regressionGravatar wm42014-08-22
| | | | | | | | | | | | | | Commit 5afc025c broke this. The reason is that mpctx->delay is updated when a new video frame is added. This value is also needed to resync audio, but it will be for the wrong PTS. They must be consistent with each other, and if they aren't, initial sync will be off by N video frames, which results at least in worse user experience. This can be reproduced by for example heavily switching between normal and 2x speed, or similar. Fix by readding the video_next_pts field (keeping its use minimal, instead of reverting the commit that removed it).
* video: get rid of video_next_pts fieldGravatar wm42014-08-22
| | | | | | Not really needed anymore. Code should be mostly equivalent. Also get rid of some other now-unused or outdated things.
* audio: add a mode to insert silence on severe A/V desyncGravatar wm42014-08-15
| | | | | | 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.
* player: use virtual time for --audio-file with ordered chaptersGravatar wm42014-08-15
| | | | | | | | | 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.
* audio: fix inverted conditionGravatar wm42014-08-06
| | | | | Recent regression. Could perhaps make gapless audio fail to work correctly.
* audio: simplify conditionGravatar wm42014-07-31
| | | | | The expression added with the previous commit (0cce8fe6) looked slightly more complicated than it has to be. The code is equivalent.
* audio: fix A/V sync in encoding modeGravatar wm42014-07-31
| | | | | | | | | | | | | | | | | | | | | 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.
* player: move video display code out of the playloopGravatar wm42014-07-30
| | | | | | | | | | 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.
* player: split seek_reset()Gravatar wm42014-07-30
| | | | | | | | 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.
* audio: better sync behavior on bogus EOFGravatar wm42014-07-30
| | | | | | | 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.
* audio: better initial sync for files where audio starts laterGravatar wm42014-07-30
| | | | | | | 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.
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