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* demux: get rid of demux_packet.new_segment fieldwm42017-10-241-3/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | The new_segment field was used to track the decoder data flow handler of timeline boundaries, which are used for ordered chapters etc. (anything that sets demuxer_desc.load_timeline). This broke seeking with the demuxer cache enabled. The demuxer is expected to set the new_segment field after every seek or segment boundary switch, so the cached packets basically contained incorrect values for this, and the decoders were not initialized correctly. Fix this by getting rid of the flag completely. Let the decoders instead compare the segment information by content, which is hopefully enough. (In theory, two segments with same information could perhaps appear in broken-ish corner cases, or in an attempt to simulate looping, and such. I preferred the simple solution over others, such as generating unique and stable segment IDs.) We still add a "segmented" field to make it explicit whether segments are used, instead of doing something silly like testing arbitrary other segment fields for validity. Cached seeking with timeline stuff is still slightly broken even with this commit: the seek logic is not aware of the overlap that segments can have, and the timestamp clamping that needs to be performed in theory to account for the fact that a packet might contain a frame that is always clipped off by segment handling. This can be fixed later.
* audio: make libaf derived code optionalwm42017-09-211-2/+1
| | | | | | | | | | | | | | | This code could not be relicensed. The intention was to write new filter code (which could handle both audio and video), but that's a bit of work. Write some code that can do audio conversion (resampling, downmixing, etc.) without the old audio filter chain code in order to speed up the LGPL relicensing. If you build with --disable-libaf, nothing in audio/filter/* is compiled in. It breaks a few features, such as --volume, --af, pitch correction on speed changes, replaygain. Most likely this adds some bugs, even if --disable-libaf is not used. (How the fuck does EOF notification work again anyway?)
* audio: introduce a new type to hold audio frameswm42017-08-165-59/+59
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
* Replace remaining avcodec_close() callswm42017-07-161-4/+1
| | | | | | | | This API isn't deprecated (yet?), but it's still inferior and harder to use than avcodec_free_context(). Leave the call only in 1 case in af_lavcac3enc.c, where we apparently seriously close and reopen the encoder for whatever reason.
* ad_spdif: minor cleanupswm42017-07-101-3/+5
| | | | | | | | Use avcodec_free_context() unstead of random other calls. Actually it was already used in the second case, but calling avcodec_close() is redundant. Don't crash if allocating a codec context fails.
* ad_lavc, vd_lavc, sd_lavc: consistently use avcodec_free_context()wm42017-07-061-7/+1
| | | | | Instead of various ad-hoc ways to achieve the same thing. (The API was added only later.)
* dec_audio, ad_lavc: change license to LGPLwm42017-06-144-22/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | All relevant authors of the current code have agreed. As always, there are the usual historical artifacts that could be mentioned. For example, there used to be a large number of decoders by various authors who were not asked, but whose code was all 100% removed. (Mostly due to FFmpeg providing all codecs.) One point of contention is that Nick Kurshev might have refactored the old audio decoder code in 2001. Basically, there are hints that it might have been done by him, such as Arpi's commit message stating that the code was imported from MPlayerXP (Nick's fork), or all the files having his name in the "maintainer" field. On the other hand, the murky history of ad.h weakens this - it could be that Arpi started this work, and Nick took it (and possibly finished it). In any case, Nick could not be reached, so there is no agreement for LGPL relicensing from him. We're changing the license anyway, and assume that his change in itself is not copyrightable. He only moved code, and in addition used the equivalent video decoder framework (done by Arpi, who agreed) as template. For example, ad_functions_s was basically vd_functions_s, which the signature of the decode callback changed to the same as audio_decode(). ad_functions_s also had a comment that said it interfaces with "video decoder drivers" (I'm fixing this comment in this commit). I verified that no additional code was added that is copyright-relevant, still in today's code, and not copied from the existing code at the time (either from the previous audio decoder code or the video framework code). What apparently matters here is that none of the old code was not written by Nick, and the authors of the old code have given his agreement, and (probably) that Nick didn't add actual new code (none that would have survived), that was not trivially based on the old one (i.e. no new copyrightable "work"). A copyright expert told me that this kind of change can be considered not relevant for copyright, so here we go. Rewriting this would end with the same code anyway, and the naming conventions can't be copyrighted.
* ad_spdif: change license to LGPLwm42017-05-211-7/+7
| | | | | All authors have agreed. (Even the main author, if you wonder about the entry in the Copyright file.)
* ad.h: change license to LGPLwm42017-05-051-7/+7
| | | | | | | | All authors have agreed. Commit 94d3170bd05 is a bit murky: Nick could not be reached, and arpi's changes were obviously inspired or copied from Nick's. However, the changed symbols were removed and do not exist anymore.
* 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 featurewm42017-02-072-0/+6
| | | | | 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".
* ad_lavc, vd_lavc: move mpv->lavc decoder parameter setup to common codewm42017-01-251-13/+5
| | | | | | | | This can be useful in other contexts. Note that we end up setting AVCodecContext.width/height instead of coded_width/coded_height now. AVCodecParameters can't set coded_width, but this is probably more correct anyway.
* build: replace some FFmpeg API checks with version checkswm42017-01-241-2/+2
| | | | | | The FFmpeg versions we support all have the APIs we were checking for. Only Libav missed them. Simplify this by explicitly checking for FFmpeg in the code, instead of trying to detect the presence of the API.
* ad_lavc: respect AV_FRAME_FLAG_DISCARDwm42017-01-241-0/+5
| | | | | | | Since we set "skip_manual", we can actually get frames with this set. Currently, only AV_PKT_FLAG_DISCARD will trigger this flag, and only mov.c sets the latter flags, so this is related to FFmpeg's half-broken mp4 edit list support.
* ad_spdif: log avformat errorswm42017-01-191-1/+3
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* ad_spdif: fix obscure cases of AC3 passthroughwm42017-01-181-7/+28
| | | | | | | | | Apparently you set the native sample rate when passing through AC3. This fixes passthrough with 44100 Hz AC3. Avoid opening a decoder for this and only open the parser. (Hopefully DTS will also support this some time in the future or so - having to open a decoder just to get the profile is dumb.)
* audio: restructure decode loopwm42017-01-114-57/+85
| | | | | | | 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 selectedwm42016-12-233-13/+48
| | | | | | | | | | | | | | 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.
* ad_lavc, vd_lavc: don't set AVCodecContext.refcounted_frameswm42016-12-181-1/+0
| | | | | This field is (or should be) deprecated, and there's no need to set it with the new API.
* ad_spdif: Fix crash when spdif muxer is not availableMichael Forney2016-12-111-0/+1
| | | | | | | Currently, if init_filter fails after lavf_ctx is allocated, uninit is called which frees lavf_ctx, but doesn't clear the pointer in spdif_ctx. So, on the next call of decode_packet, it thinks it is already initialized and uses it, resulting in a crash on my system.
* Remove compatibility thingswm42016-12-072-31/+1
| | | | | | Possible with bumped FFmpeg/Libav. These are just the simple cases.
* 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.)
* ad_lavc, vd_lavc: fix a recent libavcodec deprecation warningwm42016-10-171-1/+2
| | | | | | | | | | | | | | Both AVFrame.pts and AVFrame.pkt_pts have existed for a long time. Until now, decoders always returned the pts via the pkt_pts field, while the pts field was used for encoding and libavfilter only. Recently, pkt_pts was deprecated, and pts was switched to always carry the pts. This means we have to be careful not to accidentally use the wrong field, depending on the libavcodec version. We have to explicitly check the version numbers. Of course the version numbers are completely idiotic, because idiotically the pkg-config and library names are the same for FFmpeg and Libav, so we have to deal with this explicitly as well.
* audio: dump timestamp differencewm42016-10-021-1/+5
| | | | | Can help to analyze timestamp jitter or seeing completely bogus timestamps.
* vd_lavc, ad_lavc: set pkt_timebase, not time_basewm42016-08-291-1/+4
| | | | | | | | | These are different AVCodecContext fields. pkt_timebase is the correct one for identifying the unit of packet/frame timestamps when decoding, while time_base is for encoding. Some decoders also overwrite the time_base field with some unrelated codec metadata. pkt_timebase does not exist in Libav, so an #if is required.
* ad_lavc: actually tell decoder about the timebasewm42016-08-231-0/+1
| | | | Essentially forgotten in commit 05e4df3f.
* video/audio: always provide "proper" timestamps to libavcodecwm42016-08-191-1/+1
| | | | | | | | | | | | | | | | | | | Instead of passing through double float timestamps opaquely, pass real timestamps. Do so by always setting a valid timebase on the AVCodecContext for audio and video decoding. Specifically try not to round timestamps to a too coarse timebase, which could round off small adjustments to timestamps (such as for start time rebasing or demux_timeline). If the timebase is considered too coarse, make it finer. This gets rid of the need to do this specifically for some hardware decoding wrapper. The old method of passing through double timestamps was also a bit questionable. While libavcodec is not supposed to interpret timestamps at all if no timebase is provided, it was needlessly tricky. Also, it actually does compare them with AV_NOPTS_VALUE. This change will probably also reduce confusion in the future.
* audio: use --audio-channels=auto behavior, except on ALSAwm42016-08-041-2/+2
| | | | | | | | | | | | | | | | | | | | | | | This commit adds an --audio-channel=auto-safe mode, and makes it the default. This mode behaves like "auto" with most AOs, except with ao_alsa. The intention is to allow multichannel output by default on sane APIs. ALSA is not sane as in it's so low level that it will e.g. configure any layout over HDMI, even if the connected A/V receiver does not support it. The HDMI fuckup is of course not ALSA's fault, but other audio APIs normally isolate applications from dealing with this and require the user to globally configure the correct output layout. This will help with other AOs too. ao_lavc (encoding) is changed to the new semantics as well, because it used to force stereo (perhaps because encoding mode is supposed to produce safe files for crap devices?). Exclusive mode output on Windows might need to be adjusted accordingly, as it grants the same kind of low level access as ALSA (requires more research). In addition to the things mentioned above, the --audio-channels option is extended to accept a set of channel layouts. This is supposed to be the correct way to configure mpv ALSA multichannel output. You need to put a list of channel layouts that your A/V receiver supports.
* ad_lavc: work around braindead ffmpeg behaviorwm42016-07-011-0/+6
| | | | | | | | | | | | | | | | | | | | | The libavcodec wmapro decoder will skip some bytes at the start of the first packet and return each time. It will not return any audio data in this state. Our own code as well as libavcodec's new API handling (avcodec_send_packet() etc.) discard the PTS on the first return, which means the PTS is never known for the first packet. This results in a "Failed audio resync." message. Fixy it by remember the PTS in next_pts. This field is used only if the decoder outputs no PTS, and is updated after each frame - and thus should be safe to set. (Possibly this should be fixed in libavcodec new API handling by not setting the PTS to NOPTS as long as no real data has been output. It could even interpolate the PTS if the timebase is known.) Fixes the failure message seen in #3297.
* 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.
* ad_lavc: resume from mid-stream EOF conditions with new decode APIwm42016-06-221-0/+7
| | | | | | | | | | | | | | | | Workaround for an awful corner-case. The new decode API "locks" the decoder into the EOF state once a drain packet has been sent. The problem starts with a file containing a 0-sized packet, which is interpreted as drain packet. This should probably be changed in libavcodec (not treating 0-sized packets as drain packets with the new API) or in libavformat (discard 0-sized packets as invalid), but efforts to do so have been fruitless. Note that vd_lavc.c already does something similar, but originally for other reasons. Fixes #3106.
* ad_spdif: take care of deprecated libavcodec API usagewm42016-04-201-0/+7
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* demux_lavf, ad_lavc, ad_spdif, vd_lavc: handle FFmpeg codecpar API changewm42016-03-312-2/+5
| | | | | | | | | AVFormatContext.codec is deprecated now, and you're supposed to use AVFormatContext.codecpar instead. Handle this for all of the normal playback code. Encoding mode isn't touched.
* ad_lavc, vd_lavc: support new Libav decoding APIwm42016-03-241-0/+14
| | | | For now only found in Libav.
* ad_lavc: add codec_timebase hack toowm42016-03-241-2/+5
| | | | | vd_lavc.c had this, and soon I'll need it in ad_lavc.c too. For now it's unused.
* audio: make mp_audio_skip_samples() adjust the PTSwm42016-02-221-2/+0
| | | | Slight simplification/cleanup.
* ad_lavc: skip AVCodecContext.delay samples at beginningwm42016-02-221-0/+9
| | | | | | | | Fixes correctness_trimming_nobeeps.opus. One nasty thing is that this mechanism interferes with the container-signalled mechanism with AV_FRAME_DATA_SKIP_SAMPLES. So apply it only if that is apparently not present. It's a mess, and it's still broken in FFmpeg CLI, so I'm sure this will get fucked up later again.
* ad_lavc: make sample trimming symmetric to skippingwm42016-02-221-6/+8
| | | | | | I'm not quite sure what the FFmpeg AV_FRAME_DATA_SKIP_SAMPLES API demands here. The code so far assumed that skipping can be more than a frame, but not trimming. Extend it to trimming too.
* ad_lavc: move skipping logic out of the HAVE_AVFRAME_SKIP_SAMPLES blockwm42016-02-221-10/+13
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* ad_lavc: interpolate missing timestampswm42016-02-221-0/+9
| | | | | | | | | | This is actually already done by dec_audio.c. But if AV_FRAME_DATA_SKIP_SAMPLES is applied, this happens too late here. The problem is that this will slice off samples, and make it impossible for later code to reconstruct the timestamp properly. Missing timestamps can still happen with some demuxers, e.g. demux_mkv.c with Opus tracks. (Although libavformat interpolates these itself.)
* audio: move frame clipping to a generic functionwm42016-02-211-33/+3
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* Rewrite ordered chapters and timeline stuffwm42016-02-152-1/+75
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 fieldwm42016-02-153-6/+6
| | | | | Preparation for the timeline rewrite. The codec will be able to change, the stream header not.
* ad_lavc: fix --ad-lavc-threads rangewm42016-02-111-1/+1
| | | | | | | The code is shared with the --vd-lavc-threads option, so using 0 for auto-detection just works. But no, this is not useful. Just change it for orthogonality.
* 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 valueswm42016-02-012-16/+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 checkwm42016-01-292-14/+1
| | | | Reduces the dependency of the filter/output code on the decoder.
* audio: refactor: work towards unentangling audio decoding and filteringwm42016-01-224-148/+102
| | | | | | | | | 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.
* ad_spdif: if DTS-HD is requested, and profile unknown, use DTS-HDwm42016-01-201-1/+2
| | | | | This means there will be no loss if profile detection failed for some reason.
* audio: remove initial decoding retry limitationwm42016-01-191-3/+0
| | | | | | | | | | | | | | | Seems useless. This only helped in one case: one audio stream in the sample av_find_best_stream_fails.ts had a AC3 packets which couldn't be decoded, and for which avcodec_decode_audio4() returned 0 forever. In this specific case, playback will now not start, and you have to deselect audio manually. (If someone complains, the old behavior might be restored, but differently.) Also remove the stale "bitrate" field.
* audio: move direct packet reading from decoders to common codewm42016-01-195-29/