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* dec_video, dec_audio: remove redundant NULL-checkswm42017-02-201-2/+1
| | | | OK, they're redundant. Now stop wasting my time, coverity.
* player: add experimental stream recording featurewm42017-02-071-0/+4
| | | | | This is basically a WIP, but it can't remain in a branch forever. A warning is print when using it as it's still a bit "shaky".
* audio: restructure decode loopwm42017-01-111-12/+7
| | | | | | | Same deal as with video. Including the EOF handling. (It would be nice if this code were not duplicated, but right now we're not even close to unifying the audio and video code paths.)
* options: deprecate codec family selection in --vd/--adwm42016-12-231-7/+4
| | | | | Useless now, so get rid of it. Also affects some user-visible display things (like reported codec in use).
* audio: change how spdif codecs are selectedwm42016-12-231-3/+6
| | | | | | | | | | | | | | Remove ad_spdif from the normal codec list, and select it explicitly. One goal was to decouple this from the normal codec selection, so they're less entangled and the decoder selection code can be simplified in the far future. This means spdif codec selection is now done explicitly via select_spdif_codec(). We can also remove the weird requirements on "dts" and "dts-hd" for the --audio-spdif option, and it can just do the right thing. Now both video and audio codecs consist of a single codec family each, vd_lavc and ad_lavc.
* dec_video, dec_audio: avoid full reinit on switches to the same segmentwm42016-11-091-6/+9
| | | | | | Same deal as with the previous commit. (Unfortunately, this code is still duplicated.)
* audio: dump timestamp differencewm42016-10-021-1/+5
| | | | | Can help to analyze timestamp jitter or seeing completely bogus timestamps.
* dec_audio: fix segment boudnary switchingwm42016-06-271-3/+6
| | | | | | | | | | | | | Some bugs in this code are exposed by e.g. playing lossless audio files with --ad-lavc-threads=16. (libavcodec doesn't really support threaded audio decoding, except for lossless files.) In these cases, a major amount of audio can be buffered, which makes incorrect handling of this buffering obvious. For one, draining the decoder can take a while, so if there's a new segment, we shouldn't read audio. The segment end check was completely wrong, and used the start value.
* audio: move frame clipping to a generic functionwm42016-02-211-33/+3
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* Rewrite ordered chapters and timeline stuffwm42016-02-151-1/+73
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This uses a different method to piece segments together. The old approach basically changes to a new file (with a new start offset) any time a segment ends. This meant waiting for audio/video end on segment end, and then changing to the new segment all at once. It had a very weird impact on the playback core, and some things (like truly gapless segment transitions, or frame backstepping) just didn't work. The new approach adds the demux_timeline pseudo-demuxer, which presents an uniform packet stream from the many segments. This is pretty similar to how ordered chapters are implemented everywhere else. It also reminds of the FFmpeg concat pseudo-demuxer. The "pure" version of this approach doesn't work though. Segments can actually have different codec configurations (different extradata), and subtitles are most likely broken too. (Subtitles have multiple corner cases which break the pure stream-concatenation approach completely.) To counter this, we do two things: - Reinit the decoder with each segment. We go as far as allowing concatenating files with completely different codecs for the sake of EDL (which also uses the timeline infrastructure). A "lighter" approach would try to make use of decoder mechanism to update e.g. the extradata, but that seems fragile. - Clip decoded data to segment boundaries. This is equivalent to normal playback core mechanisms like hr-seek, but now the playback core doesn't need to care about these things. These two mechanisms are equivalent to what happened in the old implementation, except they don't happen in the playback core anymore. In other words, the playback core is completely relieved from timeline implementation details. (Which honestly is exactly what I'm trying to do here. I don't think ordered chapter behavior deserves improvement, even if it's bad - but I want to get it out from the playback core.) There is code duplication between audio and video decoder common code. This is awful and could be shareable - but this will happen later. Note that the audio path has some code to clip audio frames for the purpose of codec preroll/gapless handling, but it's not shared as sharing it would cause more pain than it would help.
* audio/video: expose codec info as separate fieldwm42016-02-151-2/+2
| | | | | Preparation for the timeline rewrite. The codec will be able to change, the stream header not.
* audio: minor simplificationwm42016-02-051-3/+0
| | | | | These fields are already deallocated by uninit_decoder(). Also remove the wrong/useless log message.
* audio/video: merge decoder return valueswm42016-02-011-11/+11
| | | | | | Will be helpful for the coming filter support. I planned on merging audio/video decoding, but this will have to wait a bit longer, so only remove the duplicate status codes.
* audio: move pts reset checkwm42016-01-291-11/+1
| | | | Reduces the dependency of the filter/output code on the decoder.
* audio: refactor: work towards unentangling audio decoding and filteringwm42016-01-221-119/+74
| | | | | | | | | Similar to the video path. dec_audio.c now handles decoding only. It also looks very similar to dec_video.c, and actually contains some of the rewritten code from it. (A further goal might be unifying the decoders, I guess.) High potential for regressions.
* audio: move direct packet reading from decoders to common codewm42016-01-191-5/+15
| | | | Another bit of preparation.
* demux: merge sh_video/sh_audio/sh_subwm42016-01-121-2/+2
| | | | | | | | | | This is mainly a refactor. I'm hoping it will make some things easier in the future due to cleanly separating codec metadata and stream metadata. Also, declare that the "codec" field can not be NULL anymore. demux.c will set it to "" if it's NULL when added. This gets rid of a corner case everything had to handle, but which rarely happened.
* player: detect audio PTS jumps, make video PTS heuristic less aggressivewm42016-01-091-1/+12
| | | | | | | | | | | | | | | | | | | | | | This is another attempt at making files with sparse video frames work better. The problem is that you generally can't know whether a jump in video timestamps is just a (very) long video frame, or a timestamp reset. Due to the existence of files with sparse video frames (new frame only every few seconds or longer), every heuristic will be arbitrary (in general, at least). But we can use the fact that if video is continuous, audio should also be continuous. Audio discontinuities can be easily detected, and if that happens, reset some of the playback state. The way the playback state is reset is rather radical (resets decoders as well), but it's just better not to cause too much obscure stuff to happen here. If the A/V sync code were to be rewritten, it should probably strictly use PTS values (not this strange time_frame/delay stuff), which would make it much easier to detect such situations and to react to them.
* dec_audio: add missing includewm42015-11-081-0/+1
| | | | Was masked by FFmpeg's terrible headers, but failed with Libav.
* audio: interpolate audio timestampswm42015-11-081-2/+12
| | | | | | | | | | | | Deal with jittering Matroska crap timestamps. This reuses the mechanism that is needed for frames without PTS, and adds a heuristic to it. If the interpolated timestamp is less than 1ms away from the real one, it might be due to Matroska timestamp rounding (or other file formats with such rounding, or files remuxed from Matroska). While there actually isn't much of a need to do this (audio PTS jittering by such a low amount doesn't negatively influence much), it helps with identifying jitter from other sources.
* audio: move PTS setting out of the decoderwm42015-11-081-3/+7
| | | | | | | Instead of requiring the decoder to set the PTS directly on the dec_audio context (including handling absence of PTS etc.), transfer the packet PTS to the decoded audio frame. Marginally simpler, and gives more control to the generic code.
* audio: fix crash on uninitwm42015-06-151-1/+1
| | | | Shit.
* audio: add --audio-spdif as new method for enabling passthroughwm42015-06-051-7/+19
| | | | | | | | | | | | | 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.)
* Update license headersMarcin Kurczewski2015-04-131-5/+4
| | | | Signed-off-by: wm4 <wm4@nowhere>
* audio: remove internal libmpg123 wrapperwm42015-03-241-5/+0
| | | | | | | We've been prefering the libavcodec mp3 decoder for half a year now. There is likely no benefit at all for using the libmpg123 one. It's just a maintenance burden, and tricks users into thinking it's a required dependency.
* player: better handling of video with no timestampswm42015-03-201-0/+3
| | | | | | | | | | | Trying to handle such video is almost worthless, but it was requested by at least 2 users. If there are no timestamps, enable byte seeking by setting ts_resets_possible. Use the video FPS (wherever it comes from) and the audio samplerate for timing. The latter was already done by making the first packet emit DTS=0; remove this again and do it "properly" in a higher level.
* audio: fix initial audio PTSwm42015-01-141-24/+25
| | | | | | | | | | Commit 5e25a3d2 broke handling of the initial frame (the one decoded with initial_audio_decode()). It didn't update the pts_offset field, leading to a shift in timestamps by one audio frame. Fix by calling the actual decode function in a single place. This requires slightly more changes than what would be necessary to fix the bug, but it also somewhat simplifies the data flow.
* audio: fix assertion failure on audio decodingwm42015-01-141-2/+2
| | | | | | | There are several cases in which a decoder may need several packets to produce some output audio. Commit 5e25a3d2 broke this. Fixes #1471.
* audio: use refcounted frames in the filter chainwm42015-01-131-24/+37
| | | | | | | | | | | | | | | | | | | The goal is switching the whole audio chain to using refcounted frames. This brings the architecture closer to FFmpeg, enables better integration with libavfilter, will reduce useless copying somewhat, and will probably allow better timestamp tracking. For now, every filter goes through a semi-awful wrapper in af_do_filter(), though. This will be fixed step by step, and the wrapper should eventually be removed. Another thing that will have to be done is improving the timestamp handling and avoiding extra copies for the AO. Some of the new code is rather similar to the video filter code (the core filter code basically just has types replaced). Such code duplication is normally very unwanted, but in this case there's probably no other choice. On the other hand, this code is pretty simple (even if somewhat tricky). Maybe there will be unified filter code in the future, but this is still far away.
* audio: make decoders output refcounted frameswm42014-11-101-119/+45
| | | | | | | | | | | | | | 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 changeswm42014-11-101-1/+0
| | | | | | | | | | 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: refactor some aspects of filter chain setupwm42014-10-021-49/+4
| | | | | | | | | | | 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: prefer libavcodec over libmpg123wm42014-09-221-1/+1
| | | | | | | | | | | | | | | | | libavcodec/libavformat now handles gapless audio better. In theory, this could be implemented with ad_mpg123 too, but since libavformat strips metadata from mp3 files and passes pure mp3 packets to the decoders only, this can't work by itself. Instead, the player must pass this metadata separately. libav* do this relatively transparently over packet "side data" (attached to AVPacket). It might also be possible to let libmpg123 handles all this by implementing it as demuxer that outputs PCM, but that would have other problems, and I think it's better to make libavformat work correctly. libmpg123 can still be used with '--ad=mpg123:mp3'. Also see issue #1101.
* Move compat/ and bstr/ directory contents somewhere elsewm42014-08-291-1/+1
| | | | | | | | | bstr.c doesn't really deserve its own directory, and compat had just a few files, most of which may as well be in osdep. There isn't really any justification for these extra directories, so get rid of them. The compat/libav.h was empty - just delete it. We changed our approach to API compatibility, and will likely not need it anymore.
* audio: change playback restart and resyncingwm42014-07-281-23/+24
| | | | | | | | | | | | | | | | | | | | | 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.
* audio: fix timestampswm42014-07-241-0/+1
| | | | | | | | | Accidentally broken in b6af44d3. For ad_lavc (and in general), the PTS was not updated correctly when filtering only parts of audio frames, and for ad_mpg123 and ad_spdif the PTS was additionally offset by the frame size. This could lead to incorrect time display, and possibly broken A/V sync.
* audio: adjust format change codewm42014-07-241-8/+9
| | | | | | Execute the format change based on whether we logically detected EOF (after filters), instead of when the decode buffer was drained. It's slightly cleaner. (The requirement of len>0 existed before.)
* audio: fix race condition in EOF codewm42014-07-241-2/+2
| | | | | | | | | | Don't return an EOF code if there's still buffered data. Also, don't call demux_stream_eof() in the playloop. There's probably nothing wrong with it, but it's cleaner not to use it. Also give AD_EOF its own value, so that a decoding error doesn't drain audio by causing an EOF condition.
* audio: cosmeticswm42014-07-241-9/+5
| | | | | | | Move a function call, which does not change semantics. Write the extra buffer sample count in a more straight-forward way; the old code was not meaningful in any way (anymore).
* audio: remove unnecessary codewm42014-07-241-3/+0
| | | | | | | | It's true that the decoder can successfully decode, but return no data (for various reasons). We don't need to handle this specially, though. We just let the decoder decode some more data. This doesn't increase the danger of an endless loop either, because audio_decode() already calls this function until enough is decoded.
* audio: move initial decode to generic codewm42014-07-211-22/+31
| | | | | | | | | | | | This commit mainly moves the initial decoding of data (done to probe the audio format) to generic code. This will make it easier to make audio decoding non-blocking in a later commit. This commit also changes how decoders return data: instead of having them write the data into a prepared buffer, they return a reference to an internal buffer (by setting dec_audio.decoded). This makes it significantly easier to handle audio format changes, since the decoders don't really need to care anymore.
* audio: remove unused metadata fieldwm42014-07-211-2/+0
| | | | | This was used for replaygain at some point, until replaygain info was passed through explicitly.
* audio: use symbolic constants instead of magic integerswm42014-07-201-5/+5
| | | | Similar to commit 26468743.
* player: hide audio/video codec and file format messageswm42014-05-311-2/+1
| | | | | None of these are very important usually. For error analysis, the plain log is useless anyway, and this information is still printed with "-v".
* options: remove deprecated --identifyMartin Herkt2014-05-041-3/+0
| | | | | | | Also remove MSGL_SMODE and friends. Note: The indent in options.rst was added to work around a bug in ReportLab that causes the PDF manual build to fail.
* player: add a --dump-stats optionwm42014-04-171-5/+6
| | | | | | | | | | | | | | | | | | | | | | | This collects statistics and other things. The option dumps raw data into a file. A script to visualize this data is included too. Litter some of the player code with calls that generate these statistics. In general, this will be helpful to debug timing dependent issues, such as A/V sync problems. Normally, one could argue that this is the task of a real profiler, but then we'd have a hard time to include extra information like audio/video PTS differences. We could also just hardcode all statistics collection and processing in the player code, but then we'd end up with something like mplayer's status line, which was cluttered and required a centralized approach (i.e. getting the data to the status line; so it was all in mplayer.c). Some players can visualize such statistics on OSD, but that sounds even more complicated. So the approach added with this commit sounds sensible. The stats-conv.py script is rather primitive at the moment and its output is semi-ugly. It uses matplotlib, so it could probably be extended to do a lot, so it's not a dead-end.
* af: add replaygain_data field to af_stream and af_instanceAlessandro Ghedini2014-04-041-0/+2
| | | | Closes #664
* af: add metadata field to af_stream and af_instanceAlessandro Ghedini2014-03-131-0/+2
| | | | | | This allows to propagate metadata information to audio filters. Closes #632
* codecs: mp_msg conversionwm42013-12-211-1/+1
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* audio: mp_msg conversionswm42013-12-211-22/+18
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* Split mpvcore/ into common/, misc/, bstr/wm42013-12-171-3/+3
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* Replace mp_tmsg, mp_dbg -> mp_msg, remove mp_gtext(), remove set_osd_tmsgwm42013-12-161-5/+5
| | | | | | | | | The tmsg stuff was for the internal gettext() based translation system, which nobody ever attempted to use and thus was removed. mp_gtext() and set_osd_tmsg() were also for this. mp_dbg was once enabled in debug mode only, but since we have log level for enabling debug messages, it seems utterly useless.
* audio: flush remaining data from the filter chain on EOFwm42013-12-051-1/+5
| | | | | | | | | | | | | | | | | This can be reproduced with: mpv short.wav -af 'lavfi="aecho=0.8:0.9:5000|6800:0.3|0.25"' An audio file that is just 1-2 seconds long should play for 8-9 seconds, which audible echo towards the end. The code assumes that when playing with AF_FILTER_FLAG_EOF, the filter will either produce output, or has all remaining data flushed. I'm not really sure whether this really works if there are multiple filters with EOF handling in the chain. To handle it correctly, af_lavfi should retry filtering if 1. EOF flag is set, 2. there were input samples, and 3. no output samples were produced. But currently it seems to work well enough anyway.
* audio/filter: change filter callback signaturewm42013-12-051-9/+8
| | | | | | | | | The new signature is actually closer to how it actually works, and someone who is not familiar to the API and how it works might make fewer fatal mistakes with the new signature than the old one. Pretty weird. Do this to sneak in a flags parameter, which will later be used to flush remaining data of at least vf_lavfi.
* Move some code from player to audio/video reset functionswm42013-11-271