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* ao_oss: fix a dumb calculationwm42017-07-211-2/+3
| | | | | | | period_size used the wrong unit, and even if the unit had been correct, was assigned the wrong value. Probably fixes #4642.
* Replace remaining avcodec_close() callswm42017-07-162-8/+2
| | | | | | | | 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.
* ao_wasapi: remove redundant / outdated commentKevin Mitchell2017-07-101-1/+0
| | | | | Where this was moved from, it made slightly more sense. Here what the comment is trying to say is already pretty obvious from the code.
* ao_waspi: use switch for handling fix_format errorsKevin Mitchell2017-07-101-5/+8
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* ao_wasapi: don't repeat format negotiation on align hackKevin Mitchell2017-07-101-4/+7
| | | | | Even if it did return a different result, the bufferFrameCount from the align hack would be wrong anyway.
* ao_wasapi: fix leak on align hackKevin Mitchell2017-07-101-1/+1
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* 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.
* audio/out: fix comment typoKevin Mitchell2017-07-091-1/+1
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* ao_wasapi: enable packed 24 bit outputKevin Mitchell2017-07-091-1/+1
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* audio/out: correct copy length in ao_read_data_convertedKevin Mitchell2017-07-091-6/+7
| | | | | | | | | | | | | Previously, the entire convert_buffer was being copied to the desination without regard to the fact that it may be packed and therefore smaller. The allocated conversion buffer was also way to big bytes * (channels * samples) ** 2 instead of bytes * channels * samples
* ao_wasapi: reorder channels and samplerates to speed up searchKevin Mitchell2017-07-091-4/+4
| | | | | | | This shouldn't affect which are chosen, but it should speed up the search by putting more common configurations earlier so that a working sample format and sample rates can be found sooner obviating the need to search them for each iteration of the outer loops.
* ao_wasapi: minor cosmetic fixesKevin Mitchell2017-07-091-3/+3
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* ao_wasapi: try correct initial formatKevin Mitchell2017-07-091-1/+4
| | | | | | | | | The loop to select the native wasapi_format for the incoming audio was not breaking correctly when it found the most desirable format. It therefore executed completely leaving the least desirable format (u8) as the choice. fixes #4582
* audio: drop AF_FORMAT_S24wm42017-07-073-44/+2
| | | | | | | | | | | This is the last sample format that was only in mpv and not in FFmpeg (except the spdif special formats). It was a huge pain, even if the removed code in af_lavrresample is pretty small after all. Note that this drops S24 from the ao_coreaudio AOs too. I'm not sure about the impact, but I expect it doesn't matter. af_fmt_change_bytes() was unused as well, so remove that too.
* ao_pcm: drop AF_FORMAT_S24 usagewm42017-07-071-1/+0
| | | | | | | I'd actually be somewhat interested in supporting this, as it could help testing the S24 conversion code. But then again it's only a pain, there's no immediate need, and it would require new options to make ao_pcm.c select this output format at all.
* ao_oss: drop AF_FORMAT_S24 usagewm42017-07-071-7/+0
| | | | Can't test / don't care.
* ao_sndio: drop AF_FORMAT_S24 usagewm42017-07-071-3/+0
| | | | | | I can't test it, so I'm dropping it without replacement. If anyone is interested in readding support, it would be done like the ao_alsa.c change.
* ao_wasapi_utils: be slightly more clever when converting channel mapwm42017-07-071-0/+5
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* ao_wasapi: drop use of AF_FORMAT_S24wm42017-07-073-117/+133
| | | | | | | | | | | Do conversion directly, using the infrastructure that was added before. This also rewrites part of format negotation, I guess. I couldn't test the format that was used for S24 - my hardware does not report support for it. So I commented it, as it could be buggy. Testing this with the wasapi_formats[] entry for 24/24 uncommented would be appreciated.
* ao_alsa: drop use of AF_FORMAT_S24wm42017-07-071-21/+56
| | | | | | | | | | | | | Instead of the infrastructure added in the previous commit to do the conversion within the AO. If this is used, and snd_pcm_status_get_avail() returns more frames than snd_pcm_write*() actually accepts, you will get some nice audio corruption. Also, this mutates the data passed via play(), which is rather fishy, but sort of doesn't matter for now. Surely this will cause unintended bugs and WTFs.
* audio/out: add helper code to do 24 bit conversion in AOwm42017-07-073-0/+129
| | | | | | | | | | | | | | | | | | | | | | | | | I plan to remove the S24 sample formats in mpv. It seems like we should still support this _somehow_ in AOs though. So the idea is to convert the data to more obscure representations (that would not be useful for filtering etc. anyway) within the AO. This commit adds helper to enable this. ao_convert_fmt is meant to provide mechanisms for this, rather than a generic audio format description (as the latter leads only to overly generic misery). The conversion also supports only cases which we think will be needed at all. The main advantage of this approach is that we get S24 out of sight, and that we could support other crazy formats (like S20). The main disadvantage is that usually S32 will be selected (if both S32 and S24 are available), and there's no user control to force S24. That doesn't really matter though, and at worst makes testing harder or will lead to unpleasant arguments with audiophiles (they'd be wrong anyway). ao_convert_fmt.pad_lsb is ignored, although if we ever find a case in which playing S32 with data in the LSBs breaks when playing it as padded 24 bit format. (For example, WAVEFORMATEXTENSIBLE recommends setting the unused bits to 0 if wValidBitsPerSample implies LSB padding.)
* 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.)
* ao_coreaudio: insane hack for passing through AC3 as float PCMwm42017-06-301-5/+65
| | | | | This uses the same hack as Kodi uses, and I suspect MPlayer/ancient mpv also did this (but didn't research that).
* ao_wasapi: UWP wrapper hack supportwm42017-06-292-7/+40
| | | | | | | | | | | UWP does not support the whole IMMDevice API. Instead, you need to use a new API (available starting from Windows 8), which is in addition not in MinGW, and extremely unpleasant to use. The wasapiuwp2.dll wrapper is a small custom MSVC DLL, which does this instead, and returns a normal IAudioClient. Before this, ao_wasapi did not initialize on UWP.
* Universal Windows Plaform (UWP) supportPedro Pombeiro2017-06-291-0/+4
| | | | | | | | libmpv only. Some things are still missing. Heavily reworked. Signed-off-by: wm4 <wm4@nowhere>
* ao_wasapi: do not use deprecated wchar functionsPedro Pombeiro2017-06-292-2/+2
| | | | These break on UWP. Based on a patch by Pedro Pombeiro.
* Avoid calling close(-1)wm42017-06-291-2/+5
| | | | | | | | | | While this is perfectly OK on Unix, it causes annoying valgrind warnings, and might be otherwise confusing to others. On Windows, the runtime can actually abort the process if this is called. push.c part taken from a patch by Pedro Pombeiro.
* ao_wasapi: remove subtly duplicated codewm42017-06-281-7/+6
| | | | Seems like this can be slightly simplified.
* ao_wasapi: remove duplicate code for creating IAudioClientwm42017-06-281-34/+24
| | | | | | | | | | | | The code accounting for the terrible AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED semantics (which MSDN claims can happen "starting with Windows 7" - so probably on Windows 10 too) duplicated the call for creating the IAudioClient. That's not great, so get rid of it. Let wasapi_thread_init() handle this. It has a retry loop anyway. This redoes device lookup and format negotiation, but potential failures due to race conditions (what if the driver decides to change behavior) shouldn't be worse than before.
* audio/out/pull: detect and log underflowswm42017-06-281-0/+20
| | | | Mostly for debugging, I guess.
* audio/out: require AO drivers to report period size and correct bufferwm42017-06-259-4/+39
| | | | | | | | | | | | | | | | Before this change, AOs could have internal alignment, and play() would not consume the trailing data if the size passed to it is not aligned. Change this to require AOs to report their alignment (via period_size), and make sure to always send aligned data. The buffer reported by get_space() now always has to be correct and reliable. If play() does not consume all data provided (which is bounded by get_space()), an error is printed. This is preparation for potential further AO changes. I casually checked alsa/lavc/null/pcm, the other AOs might or might not work.
* ao_openal: change license to LGPLwm42017-06-241-7/+7
| | | | All authors have agreed.
* chmap: remove misleading "downmix" channel layout namewm42017-06-241-1/+0
| | | | | | | | I'm not even sure when/if FFmpeg produces those. It's just confusing. If you really need this, you can still use dl-dr. I expect that most use is unintentional. Probably fixes #4545.
* ao_pulse: reorder format choiceNiklas Haas2017-06-231-2/+2
| | | | | | | | | | | | Right now, the current order pretty much means that pulse defaults to S16 for arbitrary unsupported formats, but fallback to float would make more sense since it's the easiest to convert everything to without requiring dithering, and PA will probably just internally convert things to float anyway. Also move S32 above S16, which essentially means format_maps is sorted by preference. (Although ao_pulse currently ignores this and always picks the first as a fallback)
* build: simplify OSS checks and remove changes by "bugmen0t"wm42017-06-221-6/+3
| | | | | | | | | | | | | | | | | | The user bugmen0t was apparently a shared github account with publicly available login. Thus, we can't get LGPL relicensing permission from the people who used this account. To relicense successfully, we have to remove all their changes. This commit should remove 20d1fc13, f26fb009, defbe48d. It also should remove whatever test fragments were copied from the ancient configure, as well as some configure logic (potentially that device path stuff). I think this change still preserves the most important use-cases of OSS: BSDs, and the Linux OSS emulation (the latter for testing only). According to an OSS user, the 4front checks were probably broken anyway. The SunAudio stuff was probably for (Open)Solaris, which is dead. ao_oss.c itself will remain GPL, and still contains bugmen0t changes.
* audio/format: change license to LGPLwm42017-06-202-23/+14
| | | | | | | | | | | | | | | | | | | | | | | | | Although the origins lie somewhere in libaf, which was written by "anders" and who explicitly disagreed with the LGPL relicensing, we can change the license of these files, because all code was written by "alex", who agreed with the relicensing. The only things that remain from anders' code is the AF_FORMAT_ and af_ prefixes (see e.g. 66f4e563). It was alex who redid this file and added the format identifiers we have today (507121f7). It's also nice to see that alex actually claimed copyright on format.c (221a599f). In commit efb50cab even the bitmask concept (which anders introduced with his early af_format.c code) was removed, and essentially all lines and symbols by anders were dropped. To put it into perspective: the original af_format code was for converting actual sample data and relied on OSS sample format identifiers, mpv's format.c/h provides its own sample formats, but does not do any data conversion. Remove an now inaccurate comment from format.c (it somehow even survived the typo that was present in the original commit). Also remove most of the format.c include statements - most of them are technically anders' code. We keep limits.h though.
* 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.
* encode_lavc: move from GPL 2+ to LGPL 2.1+.Rudolf Polzer2017-06-131-7/+8
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* af, vf: improvements to libavfilter bridgewm42017-05-312-1/+8
| | | | | | Add the "lavfi-" prefix (details see manpage additons). Tag the filter name as "(lavfi)" in the verbose filter list output.
* 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.)
* ao_pcm: change license to LGPLwm42017-05-201-7/+7
| | | | | | | | | | | | | | | | | | | | | All relevant authors have agreed to the relicensing. Problem cases: eca47b1a5edae: someone else gets credited for the "idea" of this change, but it doesn't seem like it was a patch (otherwise reimar would have said "patch"). Also, the associated code got essentially removed again anyway. (The option parsing was rewritten fully.) ffb529e4eb2a9: anonymous/unknown author, but the code was fully removed anyway. The struct was removed, and the modern code does explicit read/write calls. 40789473d215b: author was not contacted, but this code was removed anyway. The magic number (0x7ffff000) is still in the new code, but I don't think that is copyright relevant. c750b8ab2d3c8: the message was entirely removed.
* audio/out: change license of some core files to LGPLwm42017-05-204-28/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | All contributors of the current code have agreed. ao.c requires a "driver" entry for each audio output - we assume that if someone who didn't agree to LGPL added a line, it's fine for ao.c to be LGPL anyway. If the affected audio output is not disabled at compilation time, the resulting binary will be GPL anyway, and ootherwise the code is not included. The audio output code itself was inspired or partially copied from libao in 7a2eec4b59f4 (thus why MPlayer's audio code is named libao2). Just to be sure we got permission from Aaron Holtzman, Jack Moffitt, and Stan Seibert, who according to libao's SVN history and README are the initial author. (Something similar was done for libvo, although the commit relicensing it forgot to mention it.) 242aa6ebd40: anders mostly disagreed with the LGPL relicensing, but we got permission for this particular commit. 0ef8e555735: nick could not be reached, but the include statement was removed again anyway. 879e05a7c17: iive agreed to LGPL v3+ only, but this line of code was removed anyway, so ao_null.c can be LGPL v2.1+. 9dd8f241ac2: patch author could not be reached, but the corresponding code (old slave mode interface) was completely removed later.
* ao_wasapi: set name of event threadJames Ross-Gowan2017-05-181-0/+2
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* af_format: change license to LGPLwm42017-05-111-7/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This case is a bit weird, because MPlayer certainly also has a file named af_format.c. Both appear to have the function of converting audio data between sample formats. However, mpv's af_format.c is a rewrite, and doesn't actually do conversion by itself. It's similar to vf_format.c, and forces the generic filter chain code to insert conversion filters, instead of doing conversion explicitly. mpv's current af_format.c started out as af_force.c in d9582ad0a41. It was renamed to af_format.c in e60b8f181d, while the old af_format.c was split into two new filters. In 943c785619 the filename was changed to af_format.c as well. The new af_format.c does not contain any libaf code, except for some potentially copy & pasted skeleton and boilerplate code. (We don't account for this in per-filter file licenses, as the old libaf code has to be removed fully, at which point the filters will have to be ported to another framework, which will removed that boilerplate code.) The old filters based on af_format.c were progressively replaced and removed. Support for non-native endian and formats with signedness different from native FFmpeg was completely removed in 831d7c3c400b. The old 24 bit conversion code was removed in 552dc0d564 (made unnecessary by 5a9f817bfde8). Also list hwdec_vaglx.c as GPL-only, which doesn't have anything to do with this commit.
* af_scaletempo: change license to LGPLwm42017-05-091-7/+7
| | | | | | | | | | | | All authors have agreed. The initial commit d33703496c as well as the current code contain this line: * inspired by SoundTouch library by Olli Parviainen We assume this is about the algorithm (not the code), and the author of the original patch actually wrote all code himself.
* af_lavcac3enc: change license to LGPLwm42017-05-091-7/+7
| | | | | | | All authors have agreed. As usual with these things, this probably does not include residues from the libaf framework.
* ao_pulse, ao_rsound: change license to LGPLwm42017-05-082-14/+14
| | | | | | | | All authors have agreed. One exception is 71247a97b35, whose author was not asked, but we deem the change as trivial. (And technically it was replaced when the audio chain dropped non-native endian sample formats.)
* ao_coreaudio: change license to LGPLwm42017-05-087-49/+49
| | | | | | | | | | | | | | | | All authors have agreed to the relicensing. The code was pretty much rewritten by Stefano Pigozzi. Since the rewrite happened incrementally, and seems to include refactored portions of older code, this relicensing was done on the pre-refactor code do. The original commit adding this AO (as ao_macosx.c) credits Timothy J. Wood as original author. He was asked and agreed to LGPL. It's not entirely sure from which project this code came from, but it's probably libao. In that project, Stanley Seibert made some changes to it (who as a major developer of libao was asked just to be sure), and also Ralph Giles and Ben Hines made two small changes. The latter were not asked, but none of their code survived anyway.
* 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.
* audio/fmt-conversion: change license to LGPLwm42017-05-052-14/+14
| | | | | | | Although pretty similar to the probably unrelicensable video/fmt-conversion.c/h (basically using the same idea, but for audio), it was written by someone else. The format mapping was first added in commit ad95e046c2451.
* af: remove unused GET_VOLUME codewm42017-04-272-6/+0
| | | | The entire af code is going to be removed, but Ordnung muss sein.
* audio: fix replaygain volume scalewm42017-04-271-1/+1
| | | | | | | | The new replaygain code accidentally applied the linear gain as cubic volume level. Fix this by moving the computation of the volume scale out of the af_volume filter. (Still haven't verified whether the replaygain code works correctly.)
* options: remove remaining deprecated audio device selection optionswm42017-04-236-38/+3
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* ao_openal: kill off device listingwm42017-04-231-41/+1
| | | | | Probably helps with #4311. It surely is not the correct fix, of course. But ao_openal has no business of causing trouble anyway.
* ao_wasapi_changenotify: use %ls instead of %S for wchar_twm42017-04-201-4/+4
| | | | | %ls is C99. %S is supported by some systems, including MinGW/MSVC, but no reason to use it.
* ao_wasapi_changenotify: fix potential race conditionwm42017-04-201-8/+8
| | | | | | | IMMDeviceEnumerator_RegisterEndpointNotificationCallback() will start listening for notifications, and is the point at which callbacks can start firing. These callbacks will read the fields we set after the register calls, which is a potential race condition. Move it upwards.