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* af_lavrresample: actually free resamplerwm42013-10-201-0/+3
| | | | Fixes #304.
* command: don't allow changing volume if no audio initializedwm42013-10-122-0/+6
| | | | | | | | | | | | | | | | | | | | | | | Changing volume when audio is disabled was a feature request (github issue #215), and was introduced with commit 327a779. But trying to fix github issue #280 (volume is not correct in no-audio mode, and if audio is re-enabled, the volume set in no-audio mode isn't set), I concluded that it's not worth the trouble and the current implementation is questionable all around. (For example, you can't change the real volume in no-audio mode, even if the AO is open - this could happen with gapless audio.) It's hard to get right, and the current mixer code is already hilariously overcomplicated. (Virtually all of mixer.c is an amalgamation of various obscure corner cases.) So just remove this feature again. Note that "options/volume" and "options/mute" still can be used in idle mode to adjust the volume used next time, though these properties can't be used during playback and thus not in audio-only mode. Querying the volume still "works" in audio-only mode, though it can return bogus values.
* ad_mpeg123: support in-stream format changesThomas Orgis2013-10-061-61/+88
| | | | | | | | | git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@36461 b3059339-0415-0410-9bf9-f77b7e298cf2 Fixes playback of http://mpg123.org/test/44and22.mp3 Cherry-picked from MPlayer SVN rev. #36461, a patch by Thomas Orgis, committed by by Reimar Döffinger.
* ao_coreaudio: clear output buffer on buffer underrunStefano Pigozzi2013-10-031-0/+1
| | | | | | | | Output silence to the output buffer during underruns. This removes small occasional glitches that happen before the AUHAL is actually paused from the `audio_pause` call. Fixes #269
* audio/out: add sndio supportChristian Neukirchen2013-10-032-0/+344
| | | | Based on an earlier patch for mplayer by Alexandre Ratchov <alex@caoua.org>
* cosmetics: replace "CTRL" defines by enumswm42013-10-021-1/+3
| | | | Because why not.
* ao_coreaudio: fetch device name only for verbose log levelStefano Pigozzi2013-10-011-6/+8
| | | | | The previous code fetched the device name regardless of log level and then only printed it if log level was verbose.
* ao_jack: don’t force exact client nameMartin Herkt2013-09-301-1/+1
| | | | | | | | Trying to connect multiple mpv clients to JACK with the JackUseExactName option would fail unless the user manually specifies a unique client name. This changes the behavior to automatically generate a unique name if the requested one is already in use.
* ao_oss: add support for SNDCTL_DSP_RESET and use it when pausingPaul B Mahol2013-09-231-0/+6
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: wm4 <wm4@nowhere>
* ao_pulse: bug fix: goto the correct error handlerJohan Kiviniemi2013-09-201-2/+2
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* ao_pulse: set the property media.role=videoJohan Kiviniemi2013-09-201-2/+17
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* af: merge af_reinit() and fix_output_format()wm42013-09-202-30/+12
| | | | | | | | | | | | | | | | Calling them separately doesn't really make sense, and all existing calls to them usually combined them. One subtitle difference was that af_init() didn't wipe the filter chain if initialization of the chain itself failed, but that didn't really make sense anyway. Also remove af_init() from the code for setting balance in mixer.c. The mixer should be in the initialized state only if audio is fully initialized, so the af_init() call made no sense. Note that the filter "editing" code in command.c doesn't really do a nice job of handling errors in case recreating an _old_ (known to work) filter chain unexpectedly fails, and this obscure/rare case might be differently handled after this change.
* mixer: make struct opaquewm42013-09-204-42/+50
| | | | Also remove stray include statements from ao_alsa and ao_oss.
* mixer: restore volume with playback resumewm42013-09-202-1/+30
| | | | | | | | | Note that this is intentionally never done if the AO or softvolume is different, or if the current volume control method is thought to control system wide volume (such as ALSA) or otherwise user controllable (such as PulseAudio). The intention is to keep things robust and to avoid messing with the user's audio settings as far as possible, while still providing the ability to resume volume if it makes sense.
* mixer: don't unmute audio when raising volumewm42013-09-191-2/+0
| | | | This is rather strange behavior, away with it.
* mixer: refactor, fix some aspects of --volume handlingwm42013-09-192-68/+124
| | | | | | | | | | | | | | | | | | | | Refactor how mixer.c does volume/mute restoration and initialization. Move to handling of --volume and --mute to mixer.c. Simplify the implementation of these and hopefully fix bugs/strange behavior related to using them as file-local options (this uses a somewhat dirty trick: the option values are reverted to "auto" after initialization). Put most code related to initialization and volume restoring in probe_softvol() and restore_volume(). Having this code all in one place is less confusing. Instead of trying to detect whether to use softvol at runtime, detect it at initialization time using AOCONTROL_GET_VOLUME (same with mute, AOCONTROL_GET_MUTE). This implies we expect SET_VOLUME/SET_MUTE to work if the GET variants work. Hopefully this is always the case. This is also preparation for being able to change volume/mute settings if audio is disabled, and for allowing restoring value with playback resume.
* mixer, af_volume: use linear values instead of dBwm42013-09-192-21/+21
| | | | | | | Softvol always used a linear multiplier for volume control. This was converted to dB, and then back to linear in af_volume. Remove this non- sense. We still try to keep the command line argument to af_volume in dB, though.
* mixer: minor refactoringwm42013-09-192-26/+27
| | | | | Let struct mixer access access MPOpts to simplify some things. Rename some variables and functions. There should be no functional changes.
* af_export: fix compilation warningwm42013-09-191-2/+1
| | | | Blargh.
* Config path functions can return NULLwm42013-09-181-0/+6
| | | | | | | It's quite unlikely, but functions like mp_find_user_config_file() can return NULL, e.g. if $HOME is unset. Fix all the code that didn't check for this correctly yet.
* audio: fix playback of Musepack SV8 fileswm42013-09-011-2/+4
| | | | | | | | | | | | | | This is basically a libavcodec API oddity: it can happen that avcodec_decode_audio4() returns 0 (meaning 0 bytes were consumed). It requires you to feed the complete packet again to decode the full packet, and to successfully decode the following packets. We ignored this case with the argument that there's the danger of an endless decode loop (because nothing of that packet is apparently decoded, so it would retry forever), but change it in order to decode mpc8 files correctly. Also add some comments to explain the mess.
* audio: don't allow setting unknown formats from command linewm42013-08-261-17/+1
| | | | | | | af_str2fmt_short(), which is used by the command line option parser, allowed passing a hex number. The user could set arbitrary integers as internal audio formats, even formats which don't exist or make no sense. This is not very useful, so get rid of it.
* audio: make internal audio format 0 an invalid formatwm42013-08-263-9/+9
| | | | | | | | | | | | Having to use -1 for that is generally quite annoying. Audio formats are created from bitmasks, and it can't be excluded that 0 is not a valid format. Fix this by adjusting AF_FORMAT_I so that it is never 0. Along with AF_FORMAT_F and the special formats, all valid formats are covered and guaranteed to be non-0. It's possible that this commit will cause some regressions, as the check for invalid audio formats changes a bit.
* Some more mp_msg conversionswm42013-08-232-101/+61
| | | | | Also add a note to mp_msg.h, since it might be not clear which of the two mechanisms is preferred.
* audio/out: do some mp_msg conversionswm42013-08-228-163/+118
| | | | | | | Use the new MP_ macros for some AOs instead of mp_msg. Not all AOs are converted, and some only partially. In some cases, some additional cosmetic changes are made.
* ao: remove some leftoverswm42013-08-222-7/+0
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* core: move contents to mpvcore (2/2)Stefano Pigozzi2013-08-0635-73/+73
| | | | Followup commit. Fixes all the files references.
* Merge pull request #154 from rossy2401/wasapi-pauseDiogo Franco2013-08-051-1/+2
|\ | | | | WASAPI stops working after pause
| * ao_wasapi: don't check the audio feed while pausedJames Ross-Gowan2013-07-271-1/+2
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* | core: change speed option/property to doublewm42013-08-051-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | The --speed option and the speed property used float. Change them to double. Change the commands that manipulate the property (speed_mult/add) to double as well. Since the cycle command shares code with the add command, we change that as well. The reason for this change is that this allows better control over speed, such as stepping by semitones. Using floats is also just plain unnecessary.
* | ao_coreaudio: move to new log APIStefano Pigozzi2013-08-014-99/+101
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* | ao_coreaudio: remove useless definesStefano Pigozzi2013-08-011-3/+0
| | | | | | | | They are already defined in the header file
* | audio/out: add support for new logging APIStefano Pigozzi2013-08-012-8/+15
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* | Fix some warningsJonathan Yong2013-07-301-6/+6
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* | ao_coreaudio: use default output unit when no device is specifiedStefano Pigozzi2013-07-291-1/+3
| | | | | | | | | | | | Using the default output audio unit should provide a much better user exeperience since it changes automatically the output device based on which becomes the default one.
* | ao_coreaudio: prevent buffer underruns to output garbageStefano Pigozzi2013-07-281-1/+6
| | | | | | | | | | | | This was removed in d427b4fd. I now found a sample that causes underruns when moving to a chapter and apparently this is also a problem when taking screenshots.
* | ao_coreaudio: fix compilation on OS X 10.7Dmitry Kalinkin2013-07-281-1/+1
|/ | | | | | | | | | | | Reverts one of the changes from 18777ecf. `kAudioObjectPropertyScopeOutput` was introduced in the 10.8 SDK while `kAudioDevicePropertyScopeOutput` was moved to `AudioHardwareDeprecated.h`. Since the deprecation is silent for now (no warnings), just use the old constant. Either way, they both evaluate to 'outp', and in the 10.8 SDK the deprecated constant is defined in terms of the non-deprecated one. Fixes #155
* Fix some -Wshadow warningswm42013-07-232-5/+5
| | | | | | In general, this warning can hint to actual bugs. We don't enable it yet, because it would conflict with some unmerged code, and we should check with clang too (this commit was done by testing with gcc).
* options: make legacy hacks for AFs/VFs more explicitwm42013-07-221-0/+1
| | | | | This means that AOs/VOs with no options set do not take the legacy option parsing path, but instead report that they have no options.
* audio/out: remove options argument from init()wm42013-07-2216-31/+18
| | | | Same as with VOs in the previous commit.
* Remove subopt-helperwm42013-07-221-1/+1
| | | | Finally not used by anything anymore. Farewell.
* ao_coreaudio: fix ifdef'd conditionalStefano Pigozzi2013-07-221-1/+1
| | | | | The big endian case was not covered. Doesn't make much difference since mpv runs on Macs with x86 only, but for the sake of correctness.
* ao_coreaudio: use new option APIStefano Pigozzi2013-07-221-38/+22
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* ao_coreaudio: switch properties getters to tallocStefano Pigozzi2013-07-223-18/+19
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* ao_coreaudio: reduce verbosity of the chmapping codeStefano Pigozzi2013-07-221-2/+2
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* ao_coreaudio: revert to original device format on digital uninitStefano Pigozzi2013-07-221-3/+11
| | | | | | This is not done automatically by CoreAudio. I am told that it would a PITA to have to switch back the format manually on the device (especially if the same device is used for lpcm output).
* ao_coreaudio: refactor chmap detectionStefano Pigozzi2013-07-223-91/+96
| | | | | | b2f9e0610 introduced this functionality with code that was quite 'monolithic'. Split the functionality over several functions and ose the new macros to get array properties.
* ao_coreaudio: refactor properties codeStefano Pigozzi2013-07-224-146/+111
| | | | | | | | Introduce some macros to deal with properties. These allow to work around the limitation of CoreAudio's API being `void **` based. The macros allow to keep their client's code DRY, by not asking size and other details which can be derived by the macro itself. I have no idea why Apple didn't design their API like this in the first place.
* ao_coreaudio: move utils functions to snake_caseStefano Pigozzi2013-07-223-39/+34
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* ao_coreaudio: split ao_coreaudio_common in two filesStefano Pigozzi2013-07-225-160/+287
| | | | | | | | | * ao_coreaudio_utils: contains several utility function * ao_coreaudio_properties: contains functions to set and get audio object properties. Conflicts: audio/out/ao_coreaudio.c
* ao_coreaudio: store asbd only when selectedStefano Pigozzi2013-07-221-4/+1
| | | | | Previous code needlessly stored the input asbd before actually testing it's support against the hardware.
* ao_coreaudio: fallback to waveext on non surround inputsStefano Pigozzi2013-07-221-3/+3
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* ao_coreaudio: set channel layout based on hardware queryStefano Pigozzi2013-07-221-2/+106
| | | | this is a wip
* ao_coreaudio: fix regression in digital stream selectionStefano Pigozzi2013-07-221-1/+1
| | | | | | The condition was checked wrongly on asbd which is the input format description. This lead to the condition always being true, thus selecting lpcm streams for digital input.
* ao_coreaudio: return errors instead false in init functionsStefano Pigozzi2013-07-221-3/+3
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* ao_coreaudio: remove useless function declarationStefano Pigozzi2013-07-221-3/+0
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* ao_coreaudio: only set chmap_sel info for lpcmStefano Pigozzi2013-07-221-5/+7
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* ao_coreaudio: set channel layout bitmapStefano Pigozzi2013-07-221-1/+17
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* ao_coreaudio: move digital detection before asbd creationStefano Pigozzi2013-07-221-10/+9
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* ao_coreaudio: remove chmap selection if format is digitalStefano Pigozzi2013-07-221-1/+0
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* ao_coreaudio: remove volume multiplication by 4Stefano Pigozzi2013-07-221-2/+2
| | | | | kHALOutputParam_Volume is the linear gain so it should be at maximum 1 to keep the audio quality good. No idea why it was more than that.
* ao_coreaudio: remove device property listener on uninitStefano Pigozzi2013-07-222-12/+32
| | | | Also extract this functionality inside a function in coreaudio_common
* ao_coreaudio: print help string in one goStefano Pigozzi2013-07-221-5/+11
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* ao_coreaudio: change all ++var to var++Stefano Pigozzi2013-07-222-12/+8
| | | | | Luckily they all were inside for loops so the functionality does not actually change.
* ao_coreaudio: change private vars names to match mpv conventionsStefano Pigozzi2013-07-221-91/+101
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* ao_coreaudio: remove packetSize private variableStefano Pigozzi2013-07-221-3/+3
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* ao_coreaudio: refactor play/pauseStefano Pigozzi2013-07-221-27/+15
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* ao_coreaudio: refactor uninitStefano Pigozzi2013-07-221-17/+6
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* ao_coreaudio: remove a fixme since that seems fixedStefano Pigozzi2013-07-221-2/+0
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* ao_coreaudio: ca_msg: add trailing \n where missingStefano Pigozzi2013-07-222-6/+6
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* ao_coreaudio: refactor playStefano Pigozzi2013-07-221-16/+8
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* ao_coreaudio: extract mixmode set/unset in utility functionsStefano Pigozzi2013-07-222-46/+58
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* ao_coreaudio: move AudioStreamChangeFormat to common file and refactorStefano Pigozzi2013-07-222-86/+74
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* ao_coreaudio: extract methods to lock/unlock device for digital outputStefano Pigozzi2013-07-222-28/+26
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* ao_coreaudio: lpcm: remove buffer size calculation depending on audio unitStefano Pigozzi2013-07-221-8/+0
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* ao_coreaudio: refactor initializationStefano Pigozzi2013-07-222-396/+282
| | | | | | | | | | | | | The initialization is split more clearly between compressed and lpcm case. For the compressed case, format selection is simplified a lot and negotiation removed. The way it was written it just passed back to the core the original requested format, not what was found available on hardware. Since this is most likely useless for the compressed case, I didn't bother with this. In the future I'd like to split this AO in two one that only uses the AUHAL and the other with direct access to the hardware so that even passthrough of lcpm can be possible. This would decrease the latency, audiophiles would like that.
* ao_coreaudio: refactor print_helpStefano Pigozzi2013-07-221-26/+21
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* ao_coreaudio: split out some utility functions and refactor themStefano Pigozzi2013-07-222-310/+315
| | | | | | | Split out some utility functions that use the CoreAudio API but are not related the main task of the AOs (which is to move data correctly to the ringbuffer). These are mainly need for the verbosity of the CoreAudio API and are just obscuring the 'real' code.
* ao_coreaudio: make variable names shorterStefano Pigozzi2013-07-221-30/+30
| | | | property_address -> p_addr
* ao_coreaudio: add error check function and macroStefano Pigozzi2013-07-221-41/+68
| | | | WIP
* ao_coreaudio: dry up ca_msg and use it everywhereStefano Pigozzi2013-07-221-61/+57
| | | | | Change the ca_msg macro to pass along MSGT_AO automatically. Also use it for every output for consistency.
* ao_coreaudio: simplify digital render callbackStefano Pigozzi2013-07-221-4/+3
| | | | | It was reported that it also works by not setting the read size in the AudioBuffer (now idea how, but I will discover it later).
* ao_coreaudio: rewrite spdif render callbackStefano Pigozzi2013-07-221-39/+20
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* ao_coreaudio: simplify render callbackStefano Pigozzi2013-07-221-23/+9
| | | | | | | Read only the requested amount by the AUHAL (instead of all the buffered data). No idea what the deal is with pausing the audio units if there is no audio to play, maybe to avoid underruns of some sort. Anyway from my tests this condition never occurred so I'm removing it all.
* af_bs2b: use new option APIwm42013-07-221-84/+28
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* af_lavfi: switch to new option APIwm42013-07-221-8/+24
| | | | | This makes it actually possible to use the filter with more complicated filter graphs (such as graphs containing the "," character).
* af_scaletempo: use new option APIwm42013-07-221-84/+44
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* af_lavrresample: switch to new option APIwm42013-07-221-51/+45
| | | | | Also add a "o" suboption, which should allow fine control over libavresample.
* af_force: use new option APIwm42013-07-221-34/+21
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* audio/filter: use new option APIwm42013-07-225-163/+99
| | | | | | | | | | | | | Make the VF/VO/AO option parser available to audio filters. No audio filter uses this yet, but it's still a quite intrusive change. In particular, the commands for manipulating filters at runtime completely change. We delete the old code, and use the same infrastructure as for video filters. (This forces complete reinitialization of the filter chain, which hopefully isn't a problem for any use cases. The old code forced reinitialization too, but it could potentially allow a filter to cache things; e.g. consider loaded ladspa plugins and such.)
* af_force: add option that causes filter to fail at initializationwm42013-07-221-1/+9
| | | | This is useful for debugging.
* af: fix recovery code for filter insertion (changing volume with spdif crash)wm42013-07-221-4/+2
| | | | | | | | | | | | This code is supposed to run if dynamic filter insertion (such as when inserting a volume filter in mixer.c) fails. Then it removes all filters and recreates the default list of filters. But the code just blew up and entered an endless loop, because it removed even the sentinel in/out filters. This could happen when trying to use softvol controls while using spdif, but also other situations. Fix it by calling the correct code. Also remove these obnoxious yoda-conditions.
* audio/decode: remove macro crapwm42013-07-227-115/+61
| | | | | Declare decoders directly, instead of using the LIBAD_EXTERN macro. This is simpler (no weird magic) and more extensible.
* ao_wasapi: Make default on Windows.Diogo Franco (Kovensky)2013-07-221-3/+3
| | | | Ahead of OSS because cygwin provides OSS.
* ao_wasapi: Fix S/PDIF passthrough initDiogo Franco (Kovensky)2013-07-221-3/+4
| | | | | | | | | | MSDN tells me to multiply the samplerates by 4 (for setting up the S/PDIF signal frequency), but doesn't mention that I'm only supposed to do it on the new, NT6.1+ IEC 61937 structs. Works on my Realtek Digital Output, but as I can't connect any hardware to it I can't hear the result. Also, always ask for little-endian AC3. I'm not sure if this is supposed to be LE or NE, but Windows is LE on all platforms, so we go with LE.
* ao_wasapi: Log the passthrough format in MSGL_VDiogo Franco (Kovensky)2013-07-221-0/+5
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* ao_wasapi: Also do passthrough for AF_FORMAT_MPEG2Diogo Franco (Kovensky)2013-07-221-1/+1
| | | | That's the sample format ad_spdif uses when the source is MP3.
* ao_wasapi: Support S/PDIF passthroughDiogo Franco (Kovensky)2013-07-221-0/+45
| | | | | | | Entirely untested as this troper has no S/PDIF hardware. Refuses trying any other format if we can't use passthrough, or we would end up sending white noise at the user.
* ao_wasapi: Fix double free on uninitDiogo Franco (Kovensky)2013-07-221-2/+0
| | | | | Caused by incorrect conversion to the m_option API: since we don't allocate the state ourselves, we also don't free it ourselves.
* ao_wasapi: Support loading devices by nameDiogo Franco (Kovensky)2013-07-221-17/+65
| | | | | | | | | Do an strstr match against the device description and, if we have only a single match, take it. This works as long as the devices in the system don't change, but it's not supposed to be reliable; if one wants reliability, one uses the device ID string. Formatting.
* ao_wasapi: Don't search for devices as part of validationDiogo Franco (Kovensky)2013-07-221-23/+6
| | | | | | | | | | | This could turn valid parameters into syntax errors by the mere presence or abscence of a device (e.g. USB audio devices), so don't do that. We do validate that, if the parameter is an integer, it is not negative. We also respond to the "help" parameter, which does the same as the "list" suboption but exits after listing. Demote the validation logging to MSGL_DBG2.
* ao_wasapi: Change function macros to require semicolon after invocationDiogo Franco (Kovensky)2013-07-221-29/+29
| | | | Add semicolons where they were missing.
* ao_wasapi: Use OPT_STRING_VALIDATE for device suboptionDiogo Franco (Kovensky)2013-07-221-22/+82
| | | | | | | | | | | | | | | | | Validates by trying to pick the device using the device enumerator and aborting with out of range on failure. Refactors find_and_load_device to not use the wasapi_state; it might be called during validation. Adds missing CoInitialize/CoUninitialize calls. Remove unused variables (the SAFE_RELEASE macros keep them referenced so compiler warnings don't help finding them...). Remove the IMMDeviceEnumerator from the wasapi_state, it's only needed during initialization and initialization is now well factored enough to get rid of it. Try and connect to unplugged devices as well when using the device ID string.
* ao_wasapi: Fully convert to m_option APIDiogo Franco (Kovensky)2013-07-221-20/+10
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* ao_wasapi: Don't leak the default device's ID when listing devicesDiogo Franco (Kovensky)2013-07-221-3/+4
| | | | Also remove unused variable.
* ao_wasapi: Annotate the default device when listing devicesDiogo Franco (Kovensky)2013-07-221-4/+17
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* ao_wasapi: Refactor device listing/loadingDiogo Franco (Kovensky)2013-07-221-42/+97
| | | | | | | | | | | | Omit "{0.0.0.00000000}." on devices that start with that substring, re-add when searching for devices by ID. Log the device ID of the default device. Log the friendly name of the used device. Consistently refer to endpoints/devices as devices, as this is more consistent with mpv terminology.
* ao_wasapi0: Rename to ao_wasapiDiogo Franco (Kovensky)2013-07-222-46/+46
| | | | | Nobody knows what the 0 was for. There's no "WASAPI version 0". Just take it out.
* ao_wasapi0: Use the mix format directly in try_mix_formatDiogo Franco (Kovensky)2013-07-221-0/+2
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* ao_wasapi0: Don't complain about failed init during AO probingDiogo Franco (Kovensky)2013-07-221-8/+10
| | | | Only if the user specifically asked for ao_wasapi0.
* ao_wasapi0: Don't fail init when listing devicesDiogo Franco (Kovensky)2013-07-221-2/+0
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* ao_wasapi0: Demote "negotiation failed" message to MSGL_VDiogo Franco (Kovensky)2013-07-221-2/+2
| | | | | Could spam the console with what may be harmless in some cases. We already complain loudly if we're stuck checking this too many times.
* ao_wasapi0: Support shared mode, better format guessing methodDiogo Franco (Kovensky)2013-07-221-72/+258
| | | | | | | | | | | | | | | | | | | | | | Uses WASAPI in shared mode by default, add :exclusive flag to choose exclusive mode (duh). WASAPI works somewhat different in shared mode: the OS suggests the sample format to use, and the GetBuffer call is done slightly differently. The shared mode driver does not consume audio as fast as it notifies the thread; we need to check how much we're allowed to write. Not doing this correctly results in spamming the console with AUDCLNT_E_BUFFER_TOO_LARGE errors. When guessing formats for exclusive mode, try several sample size and sample rate combinations instead of just falling back to s16le@44100hz. If none of the rates are accepted, tries remixing >6 channels to 5.1 channels. Failing that, tries remixing to stereo. Failing everything, including the CD Red Book format, what else is left to test? Calculate buffer_block_size based on the configured channels and bytes per sample; MSDN docs say nBlockAlign is not guaranteed to be set for anything but integer PCM formats.
* ao_wasapi0: Support device enumeration and selectionDiogo Franco (Kovensky)2013-07-221-3/+215
| | | | | | | | | Adds the :list suboption to ao_wasapi0, which enumerates the audio endpoints in the system. Adds the :device=<n> suboption, which either takes an ID string (as output by list) or a device number and uses the requested device instead of the system default.
* ao_dsound: use new option APIwm42013-07-221-14/+14
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* ao_jack: use new option APIwm42013-07-221-72/+34
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* ao_oss: switch to new option APIwm42013-07-211-31/+30
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* options: remove --mixer and --mixer-channel, turn them into alsa/oss suboptswm42013-07-212-36/+19
| | | | | | These two options were supported by ALSA and OSS only. Further, their values were specific to the respective audio systems, so it doesn't make sense to keep them as top-level options.
* ao_rsound: use new option APIwm42013-07-211-27/+17
| | | | | Untested. I don't even know if this compiles. I have no clue what rsound even is.
* ao_sdl: use new option APIwm42013-07-211-29/+19
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* ao_openal: use new option APIwm42013-07-211-42/+29
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* ao_pulse: use new option APIwm42013-07-211-17/+14
| | | | Untested, but should be fine.
* options: hide encoding AO/VO in help outputwm42013-07-211-0/+1
| | | | | These can't be used manually. Encoding is enabled with -o instead, and the encoding AO/VO is selected using internal mechanisms.
* ao_alsa: use new option API (changes syntax)wm42013-07-211-66/+25
| | | | | | | | | | | | This changes how device names are handled. Before this commit, device names were mangled in strange ways to avoid clashing with the option parser syntax. "." was replaced with ",", and "=" with ":" (the user had to do the inverse to get the correct device name). The "new" option parser has multiple ways to escape option strings, so we don't need this confusing hack anymore. Add an explicit note to the manpage as well.
* ao_pcm: use new option APIwm42013-07-211-16/+12
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* ao_portaudio: use new option APIwm42013-07-211-32/+35
| | | | | This basically serves as example. All other AOs should be ported as well.
* options: use new option code for --aowm42013-07-212-91/+89
| | | | This requires completely refactoring the AO creation code too.
* ao_wasapi0: Don't starve the WASAPI thread on seeksDiogo Franco (Kovensky)2013-07-201-6/+11
| | | | | | | | | | Seeking calls thread_reset, but doesn't call thread_play. thread_reset would disable WASAPI events, but they would never get re-enabled unless the user paused and then unpaused. Keep track of whether the stream is paused or not (there already was a field for that, but it was apparently unused), and if it's not paused, call thread_play after thread_reset. Fixes mpv freezing after seeks.
* ao_wasapi0: Don't release WASAPI buffer twiceDiogo Franco (Kovensky)2013-07-201-0/+1
| | | | Would cause bogus AUDCLNT_E_OUT_OF_ORDER errors.
* ao_wasapi0: Make it compile on cygwin64Diogo Franco (Kovensky)2013-07-201-5/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | Fixes format specifies that assume windows TYPEDEFS are as long as they look like they are. Remove calls to _beginthreadex and _endthreadex, these are only present on microsoft's C runtimes. Replace by the otherwise identical CreateThread and ExitThread calls. This actually requires fixes to devicetopology.h, but the problem has been (kinda) reported to mingw-w64: <Kovensky> I see that those KSJACK* structs are supposedly declared in devicetopology.h itself, but for some reason (some of?) the decls that use them aren't seeing them? <Kovensky> ok, it seems that it expects ks.h and ksmedia.h to declare those structs, but it doesn't <Kovensky> the included files declare KDATAFORMAT, KSIDENTIFIER and LUID (and the associated pointer typedefs) <Kovensky> but everything else is essentially inside #if 0 <Kovensky> changing the #ifndef _KS_ to only include KDATAFORMAT, KSIDENTIFIER and LUID (and putting the KSJACK stuff outside that #ifndef) makes the header compile <Kovensky> it solves my immediate problem, but if that happened to begin with there's probably something more wrong with the ks headers :S
* demux_mkv: never force output sample ratewm42013-07-161-12/+5
| | | | | | | | | | | | | | Matroska has an output sample rate (OutputSamplingFrequency), which in theory should be forced instead of whatever the decoder outputs. But it appears no software (other than mplayer2 and mpv until now) actually respects this. Even worse, there were broken files around, which played correctly with (in theory) broken software, but not mplayer2/mpv. Hacks were added to our code to play these files correctly, but they didn't catch all cases. Simplify this by doing what everyone else does, and always use the decoder's sample rate instead. In particular, we try to handle all sample rate issues like libavformat's Matroska demuxer does.
* Merge branch 'remove_old_demuxers'wm42013-07-145-144/+51
|\ | | | | | | | | | | | | | | The merged branch doesn't actually just remove old demuxers, but also includes a branch of cleanups and some refactoring. Conflicts: stream/stream.c
| * ad_lavc: re-unsimplify, fix libavcodec API usagewm42013-07-111-2/+19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It turns out that some code that was removed earlier was still needed. avcodec_decode_audio4() can decode packets "partially". In that case, you have to "slice" the packet and call the decode function again. Codecs which need this are obscure and in low numbers. One sample that needs it is here: rsync://fate-suite.ffmpeg.org/fate-suite/lossless-audio/luckynight-partial.shn (This one decodes in rather small increments.) The new code is much simpler than what has been removed earlier, though. The fact that we own the packet returned by the demuxer helps a lot. Not sure what should happen if avcodec_decode_audio4() returns 0. Currently, we throw away the packet in this case. We don't want to be stuck in an endless loop (could happen if the decoder produces no output either).
| * mplayer: fix incorrect audio sync after format changeswm42013-07-111-8/+2
| | | | | | | | | | | | | | | | | | | | | | This is not directly related to the handling of format changes itself, but playing audio normally after the change. This was broken: the output byte rate was not recalculated, so audio-video sync was simply broken. Fix this by calculating the byte rate on the fly, instead of storing it in sh_audio. Format changes are relatively common (switches between stereo and 5.1 in TV recordings), so this fixes a somewhat critical bug.
| * ad_spdif: better PTS syncwm42013-07-111-1/+3
| | | | | | | | | | | | | | | | | | | | | | pts_bytes can't just be changed at the end. It must be offset to the pts value, which is reset with each packet read from the demuxer. Make sure the pts_byte field is always reset after receiving a new PTS, i.e. increment it after actually writing to the output buffer. Flush the AVFormatContext's write buffer, because otherwise the audio PTS will jump around too much: the calculation doesn't use the exact output buffer size if there's still data in the avio buffer.
| * demux: remove facility for partial packet readswm42013-07-115-73/+23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Partial packet reads were needed because the video/audio parsers were working on top of them. So it could happen that a parser read a part of a packet, and returned that to the decoder. With libavformat/libavcodec, packets are already parsed, and everything is much simpler. Most of the simplifications in ad_spdif could have been done earlier. Remove some other stuff as well, like the questionable slave mode start time reporting (could be replaced by proper code, but we don't bother). Remove the unused skip_audio_frame() functionality as well (it was used by old demuxers). Some functions become private to demux.c, like demux_fill_buffer(). Introduce new packet read functions, which have simpler semantics. Packets returned from them are owned by the caller, and all packets in the demux.c packet queue are considered unread. Remove special code that dropped subtitle packets with size 0. This used to be needed because it caused special cases in the old code.
| * ad_lavc: simplifywm42013-07-101-30/+5
| | | | | | | | | | | | | | | | | | We don't need to deal with partial packet reads, manually using an audio parser, or having to call the libavcodec decoder multiple times per packet. Actually, I'm not sure about the last point. ffplay still does this, but the ffmpeg demuxing.c example doesn't.
| * audio: remove decoder input bufferwm42013-07-102-12/+0
| | | | | | | | This was unused.
| * demux: remove audio parserwm42013-07-082-26/+5
| | | | | | | | | | | | | | | | | | | | | | The audio parser was needed only by the "old" demuxers, and demux_rawaudio. All other demuxers output already parsed packets. demux_rawaudio is usually for raw audio, so using a parser with it doesn't usually make sense. But you can also force it to read compressed formats with fixed packet sizes, in which case the parser would have been used. This use case is probably broken now, but you will be able to do the same thing with libavformat demuxers.
| * Remove old demuxerswm42013-07-071-0/+2
| | | | | | | | | | | | | | | | | | | | Delete demux_avi, demux_asf, demux_mpg, demux_ts. libavformat does better than them (except in rare corner cases), and the demuxers have a bad influence on the rest of the code. Often they don't output proper packets, and require additional audio and video parsing. Most work only in --no-correct-pts mode. Remove them to facilitate further cleanups.
* | ao_wasapi0: use new mp_ring bufferJonathan Yong2013-07-121-82/+25
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* ao_jack: allow more control about channel layoutswm42013-07-072-1/+22
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* ao_jack: increase buffer size, always round up buffer sizewm42013-07-061-2/+2
| | | | | This should help with github issue #128, which reported stuttering distorted sound with 6 channel audio, but not with 2 channels.
* ao_wasapi0: add new wasapi event mode aoJonathan Yong2013-06-182-0/+844
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* ao_dsound: fix compilationwm42013-06-161-0/+2
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* audio/out: remove ao->outburst/buffersize fieldswm42013-06-1612-73/+64
| | | | | | | | | | | | | | | The core didn't use these fields, and use of them was inconsistent accross AOs. Some didn't use them at all. Some only set them; the values were completely unused by the core. Some made full use of them. Remove these fields. In places where they are still needed, make them private AO state. Remove the --abs option. It set the buffer size for ao_oss and ao_dsound (being ignored by all other AOs), and was already marked as obsolete. If it turns out that it's still needed for ao_oss or ao_dsound, their default buffer sizes could be adjusted, and if even that doesn't help, AO suboptions could be added in these cases.
* audio/out: don't require AOs to set ao->bpswm42013-06-169-40/+28
| | | | | | | Some still do, because they use the value in other places of the init function. ao_portaudio is tricky and reads ao->bps in the stream thread, which might be started on initialization (not sure about that, but better safe than sorry).
* audio: fix af_fmt_seconds_to_bytesStefano Pigozzi2013-06-164-8/+13
| | | | Was missing samplerate
* audio/out: remove wrapper for old AOswm42013-06-163-130/+0
| | | | It's unused now.
* ao_jack: use mp_ringStefano Pigozzi2013-06-161-31/+12
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* ao_portaudio: use mp_ringStefano Pigozzi2013-06-162-57/+15
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* core: add a spsc ringbuffer implementationStefano Pigozzi2013-06-165-164/+33
| | | | | | | | | | | | | | | | | Currently every single AO was implementing it's own ringbuffer, many times with slightly different semantics. This is an attempt to fix the problem. I stole some good ideas from ao_portaudio's ringbuffer and went from there. The main difference is this one stores wpos and rpos which are absolute positions in an "infinite" buffer. To find the actual position for writing / reading just apply modulo size. The producer only modifies wpos while the consumer only modifies rpos. This makes it pretty easy to reason about and make the operations thread safe by using barriers (thread safety is guaranteed only in the Single-Producer/Single- Consumer case). Also adapted ao_coreaudio to use this ringbuffer.
* ao_coreaudio: fix output with spdifStefano Pigozzi2013-06-161-5/+4
| | | | The mute condition was inverted...
* ao_coreaudio: split ringbuffer in it's own fileStefano Pigozzi2013-06-163-73/+159
| | | | | | | | | This is hopefully the start of something good. ca_ringbuffer_read and ca_ringbuffer_write can probably cleaned up from all the NULL checks once ao_coreaudio.c gets simplyfied. Conflicts: audio/out/ao_coreaudio.c
* ao_coreaudio: move to new libao APIStefano Pigozzi2013-06-161-310/+339
| | | | | This is just a first pass and the bare minimum to make it compile and work. SPDIF is untested for lack of hardware.
* ao_coreaudio: uncrustifyStefano Pigozzi2013-06-161-491/+553
| | | | | uncrustify -l C -c TOOLS/uncrustify.cfg --no-backup --replace \ audio/out/ao_coreaudio.c
* encode_lavc strings: use new option syntaxRudolf Polzer2013-06-161-1/+1
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* options: remove --stereowm42013-06-132-14/+1
| | | | | | | Whatever this was supposed to be originally, it doesn't have much value anymore. It just forced ad_mpg123 to upmix mono to stereo by default (the audio chain can do that). As an option, it was mostly useless and misleading, so get rid of it.
* ao_oss: fix compilation on BSDwm42013-06-111-2/+3
| | | | | | This was overlooked with commit 32a898f, because OSS4 volume control is typically not available on Linux. BSD does have this feature, so the broken code broke compilation there.
* core: make options.c compile standalonewm42013-06-081-0/+2
| | | | | This also removes the split between "mplayer" and "common" opts (common opts used to be shared between mencoder and mplayer).
* ao_jack: remove global variableswm42013-06-071-71/+79
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* ao_jack: align data sizes on audio frame sizewm42013-06-071-5/+5
| | | | | | | | | | Fixes crashes when playing with certain numbers of channels. The core assumes AOs accept data aligned on channels * samplesize, and ao_jack's play() function broke that assumption: mpv: core/mplayer.c:2348: fill_audio_out_buffers: Assertion `played % unitsize == 0' failed. Fix by aligning the buffer and chunk sizes as needed.
* ao_jack: switch to new AO APIwm42013-06-071-62/+62
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* ao_jack: uncrustifywm42013-06-071-211/+236
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* ao_oss: remove duplicated format infowm42013-06-071-50/+27
| | | | | Instead of having two big switch statements to convert between two audio formats, use a single table.
* ao_oss: remove global variableswm42013-06-071-80/+104
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* ao_oss: switch to new AO APIwm42013-06-071-128/+117
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* ao_oss: uncrustifywm42013-06-071-298/+334
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* ao_openal: switch to new AO APIwm42013-06-041-52/+66
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* ao_openal: uncrustifywm42013-06-041-172/+185
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* ao_jack: add (no-)connect suboptionreimar2013-06-041-11/+17
| | | | | | | | | | | | | Add (no)connect option to ao_jack. Patch by Markus Appel [masolomaster3000 googlemail com]. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@36297 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: DOCS/man/de/mplayer.1 DOCS/man/en/mplayer.1 audio/out/ao_jack.c
* ao_dsound: remove global variableswm42013-06-041-100/+129
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* ao_dsound: switch to new AO APIwm42013-06-041-61/+70
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* ao_dsound: uncrustifywm42013-06-041-339/+375
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* core: add common function to initialize AVPacketwm42013-06-031-5/+2
| | | | | | | | | | Audio and video had their own (very similar) functions to initialize an AVPacket (ffmpeg's packet struct) from a demux_packet (mplayer's packet struct). Add a common function for these. Also use this function for sd_lavc_conv. This is actually a functional change, as some libavfilter subtitle demuxers add weird out-of-band stuff as side-data.
* Replace calls to usec_sleep()wm42013-05-265-7/+7
| | | | | | This is just dumb sed replacement to mp_sleep_us(). Also remove the now unused usec_sleep() wrapper.
* Replace all calls to GetTimer()/GetTimerMS()wm42013-05-263-16/+16
| | | | | | | | | | | | | | | | | | | | | | GetTimer() is generally replaced with mp_time_us(). Both calls return microseconds, but the latter uses int64_t, us defined to never wrap, and never returns 0 or negative values. GetTimerMS() has no direct replacement. Instead the other functions are used. For some code, switch to mp_time_sec(), which returns the time as double float value in seconds. The returned time is offset to program start time, so there is enough precision left to deliver microsecond resolution for at least 100 years. Unless it's casted to a float (or the CPU reduces precision), which is why we still use mp_time_us() out of paranoia in places where precision is clearly needed. Always switch to the correct time. The whole point of the new timer calls is that they don't wrap, and storing microseconds in unsigned int variables would negate this. In some cases, remove wrap-around handling for time values.
* ao_alsa: always unset ALSA error handler, cleanup on init errorwm42013-05-261-8/+9
| | | | | | | | | | | | | | | | The ALSA device was not closed when initialization failed. The ALSA error handler (set with snd_lib_error_set_handler()) was not unset when closing ao_alsa. If this is not done, the handler will still be called when other libraries using ALSA cause errors, even though ao_alsa was long closed. Since these messages were prefixed with "[AO_ALSA]", they were misleading and implying ao_alsa was still used. For some reason, our error handler is still called even after doing snd_lib_error_set_handler(NULL), which should be impossible. Checking with the debuggers, inserting printf(), as well as the alsa-lib source code all suggest our error handler should not be called, but it still happens. It's a complete mystery.
* af_lavfi: add libavfilter bridgewm42013-05-232-0/+310
| | | | | | | | | | | | | | | | | | | | | Mostly copied from vf_lavfi. The parts that could be shared are minor, because most code is about setting up audio and video, which are too different. This won't work with Libav. I used ffplay.c as guide, and noticed too late that their setup methods are incompatible with Libav's. Trying to make it work with both would be too much effort. The configure test for av_opt_set_int_list() should disable af_lavfi gracefully when compiling with Libav. Due to option parser chaos, you currently can't have a "," as part of the filter graph string - not even with quoting or escaping. This will probably be fixed later. The audio filter chain is not PTS aware. So we have to do some hacks to make up a fake PTS, and we have to map the output PTS back to the filter chain's method of tracking PTS changes and buffering, by adjusting af->delay.
* chmap: fix oddity due to ambiguous 6.1 ffmpeg channel layoutwm42013-05-131-2/+3
| | | | | | | | FFmpeg (as well as Libav) have two layouts called "6.1": AV_CH_LAYOUT_6POINT1 and AV_CH_LAYOUT_6POINT1_BACK. We call them "6.1" and "6.1(back)". Change the default layout for 7 channels as well to return the same layout as av_get_default_channel_layout(). (Looks a bit questionable, but for now it's better to follow FFmpeg.)
* audio: fix ALSA 4 channel surround outputwm42013-05-132-2/+4
| | | | | | | | | It turns out that ALSA's 4 channel layout is different from mpv's and ffmpeg's 4.0 layout. Thus trying to do 4 channel output led to incorrect remixing via lib{av,sw}resample. Fix the default layouts for the internal filter chain as well, although I'm not sure if it matters at all.
* af_lavrresample: fix inverted conditionwm42013-05-131-1/+1
| | | | | This was added with the previous commit. It likely broke some obscure special-cases, which (hopefully) do not happen with normal playback.
* audio: fix compilation with older libavresample versionswm42013-05-133-1/+68
| | | | | | | | | | | | | | The libavresample version of the current Libav stable release lacks the avresample_set_channel_mapping() function. (FFmpeg's libswresample seems to be fine, because they added swr_set_channel_mapping() first.) Add a cheap/slow workaround to do channel reordering on our own. We don't use the recently removed MPlayer code (see commit 586b75a), because that is not generic enough. The functionality should be the same as with full-featured libavresample, and any differences are bugs. It's probably slower, though.
* ao_coreaudio: fix switched parameterswm42013-05-121-1/+1
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* Merge branch 'audio_changes'wm42013-05-1262-3498/+3143
|\ | | | | | | | | Conflicts: audio/out/ao_lavc.c
| * af: improve filter chain setup retry limitwm42013-05-121-1/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | af_reinit() is responsible for inserting automatic conversion filters for channel remixing, format conversion, and resampling. We don't require that a single filter can do all these (even though af_lavrresample does nearly all of this, sometimes af_format has to be used instead for format conversions). This makes setting up the chain more complicated, and a way is needed to prevent endless appending of conversion filters if a conversion is not possible. Until now, this used a stupidly simple yet robust static retry limit to detect failure. This is perfectly fine, and the limit (20) was good enough to handle about ~5 filters. But with more filters, and if each filter requires 3 additional conversion filters, this would fail. So raise the limit to 4 retries per filter. This is still stupidly simple and robust, but won't arbitrarily fail if the filter count is too large.
| * audio: add double sample formatwm42013-05-123-14/+12
| | | | | | | | | | | | To make this easier, get rid of the direct mapping of the AF_FORMAT_BITS_MASK bit field to number of bytes. This way we can throw away the unused AF_FORMAT_48BIT and don't have to add ..._56BIT.
| * ao_alsa: set fallback if format unknownwm42013-05-121-4/+6
| | | | | | | | | | | | The snd_pcm_hw_params_test_format() call actually crashes in alsa-lib if called with SND_PCM_FORMAT_UNKNOWN, so the already existing fallback code won't work in this case.
| * audio/out: channel map selectionwm42013-05-1214-68/+114
| | | | | | | | | | | | | | | | | | Make all AOs use what has been introduced in the previous commit. Note that even AOs which can handle all possible layouts (like ao_null) use the new functions. This might be important if in the future ao_select_champ() possibly honors global user options about downmixing and so on.
| * audio: add channel map selection functionwm42013-05-127-20/+297
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The point is selecting a minimal fallback. The AOs will call this through the AO API, so it will be possible to add options affecting the general channel layout selection. It provides the following mechanism to AOs: - forcing the correct channel order - downmixing to stereo if no layout is available - allow 5.1 <-> 5.1(side) fallback - handling "unknown" channel layouts This is quite weak and lots of code/complexity for little gain. All AOs already made sure the channel order was correct, and the fallback is of little value, and could perhaps be done in the frontend instead, like stereo downmixing with --channels=2 is handled. But I'm not really sure how this stuff should _really_ work, and the new code will hopefully provides enough flexibility to make radical changes to channel layout negotiation easier.
| * ao_pulse: move format setup codewm42013-05-121-27/+27
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| * af_lavrresample: avoid channel reordering with unknown layoutswm42013-05-121-8/+24
| | | | | | | | | | | | | | If one of the input or output is an unknown layout, but the other is known, it can still happen that channels are remixed randomly. Avoid this by forcing default layouts in this case. (Doesn't work if the channel counts are different.)
| * ao_openal: use channel map instead of ALSA fixed layoutwm42013-05-121-10/+31
| | | | | | | | | | | | | | | | | | Now mpv's channel map is used to map each channel to a speaker. This allows in theory for playback of any layout for which ao_openal actually has a speaker defined. Also add the back-center (BC) speaker, which allows playback of 6.0 audio. Enabling more layouts by adding other speakers would be possible, but I'm not sure about the speaker positions.
| * audio/filters: add af_forcewm42013-05-124-0/+155
| | | | | | | | | | Its main purpose is for testing in case channel layout stuff breaks, in particular in connection with old audio filters.
| * ao: remove ao_driver.is_new fieldwm42013-05-129-9/+0
| | | | | | | | Is unused, is completely pointless.
| * ao_alsa: remove global variableswm42013-05-121-110/+116
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| * ao_alsa: switch to new AO APIwm42013-05-121-39/+54
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| * af_ladspa: code cleanupeng2013-05-121-15/+17
| | | | | | | | | | | | Cleanup based on results from cppcheck-1.59 Reduce the scope of several variables Fix memory leak
| * audio: print channel map additionally to channel count on terminalwm42013-05-124-19/+31
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| * ao_alsa: map to exact channel layoutwm42013-05-121-38/+48
| | | | | | | | | | | | | | This allows supporting 5 channel audio (which can be eother 5.0 or 4.1). Fallback doesn't work yet. It will do nonsense if the channel layout doesn't match perfectly, even though it's similar.
| * ao_alsa: move format lookup into separate functionwm42013-05-121-58/+36
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| * ao_alsa: more reformatwm42013-05-121-118/+113
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| * af: print filter chain info on errorwm42013-05-121-15/+16
| | | | | | | | | | The filter chain was only visible with -v. Always print it if the filter chain could not be configured.
| * ao_alsa: cosmetics, macro-fy error reportingwm42013-05-121-321/+181
| | | | | | | | | | | | | | | | Add a CHECK_ALSA_ERROR macro to report ALSA errors. This is similar to what vo_vdpau does. This removes lots of boiler plate, it almost gives me the feeling the ao_alsa initialization code is now readable. This change is squashed with the reformatting, because both changes are just as noisy and useless.
| * ao_alsa: uncrustifywm42013-05-121-623/+717
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| * ad_lavc: force channel layout pass-through with demux_rawaudiowm42013-05-121-2/+11
| | | | | | | | | | | | | | | | Using demux_rawaudio and the --rawaudio-channels option is useful for testing channel map stuff. The libavcodec PCM decoder normalizes the channel map to ffmpeg order, though. Prevent this by forcing the original channel map when using the mp-pcm pseudo decoder entry (used by demux_rawaudio and stream/tv.c only).
| * ao_coreaudio: switch to WAVEEXT channel orderwm42013-05-121-2/+1
| | | | | | | | | | This used ALSA order, which was not correct. Most likely this has been wrong since forever.
| * ao_pulse: try to set correct channel layoutwm42013-05-121-2/+54
| | | | | | | | | | | | | | | | | | | | | | | | Like most other AOs, ao_pulse set the channel count only, always using a default layout. Try to set the exact layout. For this, we need a big lookup table to map waveex/lavc/mpv speaker position to PulseAudio's, since PA_CHANNEL_POSITION_ is apparently not compatible to waveext, and I haven't seen any API functions that would help mapping them. Completely untested. (Let's leave that to someone else...)
| * core: use channel map on demuxer level toowm42013-05-124-21/+34
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This helps passing the channel layout correctly from decoder to audio filter chain. (Because that part "reuses" the demuxer level codec parameters, which is very disgusting.) Note that ffmpeg stuff already passed the channel layout via mp_copy_lav_codec_headers(). So other than easier dealing with the demuxer/decoder parameters mess, there's no real advantage to doing this. Make the --channels option accept a channel map. Since simple numbers map to standard layouts with the given number of channels, this is downwards compatible. Likewise for demux_rawaudio.
| * reorder_ch: remove old channel reorder functionswm42013-05-124-1497/+0
| | | | | | | | | | | | | | This is done in af_lavrresample now, and as part of format negotiation. Also remove the remaining reorder_channel calls. They were redundant and did nothing.
| * audio: let libavresample do channel reorderingwm42013-05-121-3/+49
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| * af_lavrresample: context is always allocated herewm42013-05-121-2/+1
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| * audio/out: switch to channel mapwm42013-05-1219-235/+158
| | | | | | | | | | | | This actually breaks audio for 5/6/8 channels. There's no reordering done yet. The actual reordering will be done inside of af_lavrresample and has to be made part of the format negotiation.
| * af_pan: set unknown channel layout for outputwm42013-05-121-2/+11
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| * af: use mp_chmap for mp_audio, include channel map in format negotiationwm42013-05-1212-57/+64
| | | | | | | | | | Now af_lavrresample pretends to reorder the channels, although it doesn't yet, and nothing sets non-standard layouts either.
| * audio: add some setters for mp_audio, and require filters to use themwm42013-05-1228-178/+166
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | mp_audio has some redundant fields. Setters like mp_audio_set_format() initialize these properly. Also move the mp_audio struct to a the file audio.c. We can remove a mysterious line of code from af.c: in.format |= af_bits2fmt(in.bps * 8); I'm not sure if this was ever actually needed, or if it was some kind of "make it work" quick-fix that works against the way things were supposed to work. All filters etc. now set the format correctly, so if there ever was a need for this code, it's definitely gone.
| * audio: add channel map APIwm42013-05-123-3/+613
| | | | | | | | | | | | | | Unused, will be used in the following commits. Let chmap.h define the number of maximum channels, because that is most convenient.
| * audio/filter: remove unused AF_CONTROLswm42013-05-127-213/+1
| | | | | | | | Was unused, has never been used.
| * options: add option to prevent decoder audio downmixingwm42013-04-131-3/+24
| | | | | | | | Also rename --a52drc to --ad-lavc-ac3drc, and add --ad-lavc-o.
| * af: fix negotiation endless loopwm42013-04-131-3/+2
| | | | | | | | | | | | | | Yeah... ok. Can be reproduced by having AF_CONTROL_CHANNELS not really set the correct channel map.
| * af: streamline format negotiationwm42013-04-133-167/+210
| | | | | | | | | | | | | | | | | | | | | | | | | | Add dummy input and output filters to remove special cases in the format negotiation code (af_fix_format_conversion() etc.). The output of the filter chain is now negotiated in exactly the same way as normal filters. Negotiate setting the sample rate in the same way as other audio parameters. As a side effect, the resampler is inserted at the start of the filter chain instead of the end, but that shouldn't matter much, especially since conversion and channel mixing are conflated into the same filter (due to libavresample's API).
| * af_lavrresample: add no-detach suboptionwm42013-04-131-1/+6
| | | | | | | | | | Normally, af_lavrresample detaches itself immediately if the input and output audio parameters are the same. no-detach prevents this.
| * options: remove --af-advwm42013-04-133-23/+1
| | | | | | | | | | | | | | Anything this option did has been removed in the preceding 3 commits. Note that even though these options sounded like a good idea (like setting accuracy vs. speed tradeoffs), they were not really properly implemented.
| * af: remove accuracy optionwm42013-04-131-15/+4
| | | | | | | | | | | | | | | | All this option did was deciding whether the resample filter was to be insert at the beginning or end of the filter chain. Always do what the option set for accuracy did. I doubt it makes much of a difference. libavresample does most things in just one go anyway, so it won't matter.
| * af: remove force optionwm42013-04-131-64/+49
| | | | | | | | | | | | Dangerous and misleading. If it turns out that this is actually needed to make certain setups work right, it should be added back in a better way (in a way it doesn't cause random crashes).
| * audio: remove float processing optionwm42013-04-135-30/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The only thing this option did was changing the behavior of af_volume. The option decided what sample format af_volume would use, but only if the sample format was not already float. If the option was set, it would default to float, otherwise to S16. Remove use of the option and all associated code, and make af_volume always use float (unless a af_volume specific sub-option is set). Silence maximum value tracking. This message is printed when the filter is destroyed, and it's slightly annoying. Was enabled due to enabling float by default.
| * audio: switch to libavcodec channel order, use libavresample for mixingwm42013-04-135-44/+44
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Switch the internal channel order to libavcodec's. If the channel number mismatches at some point, use libavresample for up- or downmixing. Remove the old af_pan automatic downmixing. The libavcodec channel order should be equivalent to WAVEFORMATEX order, at least nowadays. reorder_ch.h assumes that WAVEFORMATEX and libavcodec might be different, but all defined channels have the same mappings. Remove the downmixing with af_pan as well as the channel conversion with af_channels from af.c, and prefer af_lavrresample for this. The automatic downmixing behavior should be the same as before (if the --channels option is set to 2, which is the default, the audio output is forced to 2 channels, and libavresample does all downmixing). Note that mpv still can't do channel layouts. It will pick the default channel layout according to the channel count. This will be fixed later by passing down the channel layout as well. af_hrtf depends on the order of the input channels, so reorder to ALSA (for which this code was written). This is better than changing the filter code, which is more risky. ao_pulse can accept waveext order directly, so set that as channel mapping.
| * ao_alsa: reorder channels from internal to alsa orderwm42013-04-131-0/+6
| | | | | | | | | | Currently, internal and alsa order are exactly the same, so this will do absolutely nothing.
| * af: simplificationwm42013-04-131-35/+20
| | | | | | | | | | | | | | | | | | If format negotiation fails, and additional filters are inserted to fix this, don't try to reinitialize the filter immediately. Instead, correct the audio format, and let the caller retry. Add a retry counter to af_reinit() to ensure that misbehaving filters can't put the format negotiation into an endless loop.
| * af: factor channel filter insertionwm42013-04-131-30/+45
| | | | | | | | Do this just like it has been done for the format filter.
| * af: use af_lavrresample for format conversions, if possiblewm42013-04-134-45/+111
| | | | | | | | | | | | | | | | | | | | | | | | | | Refactor to remove the duplicated format filter insertion code. Allow other format converting filters to be inserted on format mismatches. af_info.test_conversion checks whether conversion between two formats would work with the given filter; do this to avoid having to insert multiple conversion filters at once and such things. (Although this isn't ideal: what if we want to avoid af_format for some conversions? What if we want to split af_format in endian-swapping filters etc.?) Prefer af_lavrresample for conversions that it supports natively, otherwise let af_format handle the full conversion.
| * af: remove automatically inserted filters on full reinitwm42013-04-132-31/+44
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Make sure automatically inserted filters are removed on full reinit (they are re-added later if they are really needed). Automatically inserted filters were never explicitly removed, instead, it was expected that redundant conversion filters detach themselves. This didn't work if there were several chained format conversion filters, e.g. s16le->floatle->s16le, which could result from repeated filter insertion and removal. (format filters detach only if input format and output format are the same.) Further, the dummy filter (which exists only because af.c can't handle an empty filter chain for some reason) could introduce bad conversions due to how the format negotiation works. Change the code so that the dummy filter never takes part on format negotiation. (It would be better to fix format negotiation, but that would be much more complicated and would involving fixing all filters.) Simplify af_reinit() and remove the start audio filter parameter. This means format negotiation and filter initialization is run more often, but should be harmless.
| * af_lavrresample: allow other ffmpeg sample formats for input/outputwm42013-04-131-17/+48
| | | | | | | | | | | | | | The format was locked to s16. Extend it to accept all other ffmpeg sample formats, and even allow different in- and output formats. The generic filter code will still insert af_format on format mismatches, though.
| * audio/filter: replace pointless memcpys with assignmentswm42013-04-134-17/+10
| | | | | | | | | | | | The change in af_scaletempo actually fixes a memory leak. af->data contained a pointer to an allocated buffer, which was overwritten during format negotiation. Set the format explicitly instead.
| * af: uncrustifywm42013-04-132-466/+496
| |
| * ad_lavc: use fmt-conversion to map sample formatsStefano Pigozzi2013-04-133-18/+95
| |
* | OSX: use native Cocoa's event loopStefano Pigozzi2013-05-121-5/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Schedule mpv's playloop as a high frequency timer inside the main Cocoa event loop. This has the benefit to allow accessing menus as well as resizing the window without the playback being blocked and allows to remove countless hacks from the code that involved manually pumping the event loop as well simulating manually some of the Cocoa default behaviours. A huge improvement consists in removing NSApplicationLoad. This is a C function defined in the Cocoa header and implements a minimal OSX application under ther hood so that you can use the Cocoa GUI toolkit from C/C++ without having to respect the Cocoa standards in terms of application initialization. This was bad because the behaviour implemented by NSApplicationLoad was hard to customize and had several gotchas especially in the menu department. mpv was changed to be just a nib-less application. All the Cocoa part is still generated in code but the event handling is now not dissimilar to what is present in a stock Mac application. As a part of reviewing the initialization process, I also removed all of `osdep/macosx_finder_args`. The useful parts of the code were moved to `osdep/macosx_appication` which has the broaded responsibility of managing the full lifecycle of the Cocoa application. By consequence the `--enable-macosx-finder` configure switch was killed as well, as this feature is always enabled. Another change the users will notice is that when using a bundle the `--quiet` option will be inserted much earlier in the initializaion process. This results in mpv not spamming mpv.log anymore with all the initialization outputs.
* | encoding: fix final audio frame syncRudolf Polzer2013-04-281-10/+22
| | | | | | | | | | When --ocopyts was used, the final audio frame got improper pts. Fixed by now using the play() logic to play the final frame too.
* | audio/decode: remove vararg from ad_control()wm42013-04-125-5/+5
|/ | | | | This was unused and dumb. Ancient MPlayer used varargs instead of void* arguments for control() functions, and this was the last leftover.
* ao_jack: fix deprecation warningStefano Pigozzi2013-04-121-2/+5
| | | | | jack_port_get_total_latency is deprecated: use the "new" API based on jack_port_get_latency_range instead.
* mplayer: keep volume persistent, even when using --volumewm42013-04-102-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | Consider: mpv --volume 10 file1.mkv file2.mkv Before this commit, the volume was reset to 10 when playing file2.mkv. This was inconsistent to most other options. E.g. --brightness is a rather similar case. In general, settings should never be reset when playing the next file, unless the option was explicitly marked file-local. This commit corrects the behavior of the --volume and --mute options. File local --volume still works as expected: mpv --{ --volume 10 file1.mkv file2.mkv --} This sets the volume always to 10 on playback start. Move the m_config_leave_file_local() call down so that the mixer code in uninit_player() can set the option volume and mute variables without overwriting the global option values. Another subtle issue is that we don't want to set volume if there's no need to, which is why the user_set_volume/mute fields are introduced. This is important because setting the volume might change the system volume depending on other options.
* ao_dsound: add missing includeKovensky2013-03-231-0/+1
| | | | libavutil/common.h is needed for FF_ARRAY_ELEMS.
* af_lavrresample: add new resampling filter to replace the old onesStefano Pigozzi2013-03-135-795/+280
| | | | | | | | | | Remove `af_resample` and `af_lavcresample`. The former is a mess while the latter uses an API that was long deprecated in libavcodec and is now removed. `af_lavrresample` rougly has the same features and structure of `af_lavcresample`. libswresample fallback by wm4.
* Prefix CODEC_ID_ with AV_wm42013-03-132-15/+15
| | | | | | | | | The old names have been deprecated a while ago, but were needed for supporting older ffmpeg/libav versions. The deprecated identifiers have been removed from recent Libav and FFmpeg git. This change breaks compatibility with Libav 0.8.x and equivalent FFmpeg releases.
* af_lavcac3enc: switch to avcodec_encode_audio2()wm42013-03-131-45/+71
| | | | | | | | avcodec_encode_audio() was deprecated, and was finally removed from Libav and FFmpeg git. This appears to work. I get heavy A/V desync with -ao alsa and -ao pcm, but this was already so before this change.
* Add a --dtshd optionWessel Dankers2013-03-041-13/+22
| | | | | | | | | | The spdif decoder was hardcoded to assume that the spdif output is capable of accepting high (>1.5Mbps) bitrates. While this is true for modern HDMI spdif interfaces, the original coax/toslink system cannot deal with this and will fail to work. This patch adds an option --dtshd which can be enabled if you use a DTS-capable receiver behind a HDMI link.
* Rename af_volnorm to af_drcMartin2013-02-122-20/+20
| | | | | | The previous name of this filter was misleading, because it doesn’t actually normalize volume levels. What it does is closer to performing low-quality dynamic range compression, hence it is now called af_drc.
* demux_lavf, ad_lavc, vd_lavc: pass codec header data directlywm42013-02-101-8/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | Instead of putting codec header data into WAVEFORMATEX and BITMAPINFOHEADER, pass it directly via AVCodecContext. To do this, we add mp_copy_lav_codec_headers(), which copies the codec header data from one AVCodecContext to another (originally, the plan was to use avcodec_copy_context() for this, but it looks like this would turn decoder initialization into an even worse mess). Get rid of the silly CodecID <-> codec_tag mapping. This was originally needed for codecs.conf: codec tags were used to identify codecs, but libavformat didn't always return useful codec tags (different file formats can have different, overlapping tag numbers). Since we don't go through WAVEFORMATEX etc. and pass all header data directly via AVCodecContext, we can be absolutely sure that the codec tag mapping is not needed anymore. Note that this also destroys the "standard" MPlayer method of exporting codec header data. WAVEFORMATEX and BITMAPINFOHEADER made sure that other non-libavcodec decoders could be initialized. However, all these decoders have been removed, so this is just cruft full of old hacks that are not needed anymore. There's still ad_spdif and ad_mpg123, bu neither of these need codec header data. Should we ever add non-libavcodec decoders, better data structures without the past hacks could be added to export the headers.
* demux_lavf, ad_lavc, vd_lavc: refactor, cleanupwm42013-02-101-24/+18
| | | | | | | | | Rearrange some code to make it easier readable. Remove some dead code, and stop printing AVI headers in demux_lavf. (These are not actual AVI headers, just for internal use.) There should be no functional changes, other than reducing output in verbose mode.
* core: redo how codecs are mapped, remove codecs.confwm42013-02-109-250/+148
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Use codec names instead of FourCCs to identify codecs. Rewrite how codecs are selected and initialized. Now each decoder exports a list of decoders (and the codec it supports) via add_decoders(). The order matters, and the first decoder for a given decoder is preferred over the other decoders. E.g. all ad_mpg123 decoders are preferred over ad_lavc, because it comes first in the mpcodecs_ad_drivers array. Likewise, decoders within ad_lavc that are enumerated first by libavcodec (using av_codec_next()) are preferred. (This is actually critical to select h264 software decoding by default instead of vdpau. libavcodec and ffmpeg/avconv use the same method to select decoders by default, so we hope this is sane.) The codec names follow libavcodec's codec names as defined by AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders have names different from the canonical codec name. The AVCodecDescriptor API is relatively new, so we need a compatibility layer for older libavcodec versions for codec names that are referenced internally, and which are different from the decoder name. (Add a configure check for that, because checking versions is getting way too messy.) demux/codec_tags.c is generated from the former codecs.conf (minus "special" decoders like vdpau, and excluding the mappings that are the same as the mappings libavformat's exported RIFF tables). It contains all the mappings from FourCCs to codec name. This is needed for demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the codec as determined by libavformat, while the other demuxers have to do this on their own, using the mp_set_audio/video_codec_from_tag() functions. Note that the sh_audio/video->format members don't uniquely identify the codec anymore, and sh->codec takes over this role. Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which provide cover the functionality of the removed switched. Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure container/video combinations (e.g. the sample Film_200_zygo_pro.mov) are played flipped. ffplay/avplay doesn't handle this properly either, so we don't care and blame ffmeg/libav instead.
* dec_audio: uncrustifywm42013-02-092-193/+203
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* audio/out, video/out: hide encoding VO/AOwm42013-02-063-8/+19
| | | | | | mpv -ao help and mpv -vo help shouldn't show the encoding outputs (named "lavc" on both cases). Also make it impossible to select these manually when not encoding.
* audio/out: prefer ao_dsound over ao_portaudiowm42013-02-061-3/+3
| | | | | | On Linux, ao_portaudio has weird freezing issues (possibly specific to the ALSA backend, though). Also ao_dsound is more likely to get multi- channel audio output right, and ao_portaudio probably mangles these.
* mixer: prefer AO softvol control over volume filterwm42013-02-062-2/+3
| | | | | | | | | | | | | | | This partially reverts earlier decisions, when I thought it would always be better to prefer the audio volume filter over the AO's, because the AO's relies on the underlying audio-API, which could be broken or exhibit unusual behavior (like it happened with ao_dsound). However, since the audio buffer can be quite large (500 ms), and we don't attempt to flush & refilter the audio on volume changes, always prefer AO volume control (as long as the AO mixer doesn't control the system mixer). Also document what the mixer.c related AO fields mean (hopefully not too brief).
* ao_dsound: support 6.1 and 7.1 channel configurationswm42013-02-061-30/+43
| | | | | | | Instead of doing the channel reordering manually, use the existing support in reorder_ch.c. Untested.
* ao_coreaudio: use 0 as timeout for CFRunLoopRunInModeMad Fish2013-01-201-1/+1
| | | | | | | Handle all pending events and exit instead of waiting. When there are lots of input events (for example, scrolling with trackpad), timeout can add up to make a huge frame delay. In my tests, if I scroll fast enough, that loop would never exit.
* Replace strsep() useswm42013-01-131-2/+7
| | | | | | This function sucks and apparently is not very portable (at least on mingw, the configure check fails). Also remove the emulation of that function from osdep/strsep*, and remove the configure check.
* mixer: keep fractional part of volume settingUoti Urpala2013-01-131-2/+2
| | | | | | | | mixer_setvolume() accepts float values for volume, but used the integer function av_clip() to limit range, losing the fractional part as a side effect. Change the code to use av_clipf() instead. For most uses this shouldn't make any real difference; actual AO volume settings may not have that much precision anyway.
* af_volnorm: fix output range with float inputUoti Urpala2013-01-131-3/+3
| | | | | | | | | af_volnorm can process either int16_t or float audio data. The float version used 0 to INT_MAX as full value range, when it should be 0 to 1. This effectively disabled the filter (due to all input being considered to fall in the silence range). Fix. Reported by Tobias Jacobi <liquid.acid@gmx.net>.
* ao_alsa: do not call snd_pcm_delay() when pausedwm42013-01-061-0/+8
| | | | | | | | | | | | | | | | | | | | | | This causes trouble when a hw device is used: pcm_hw.c:514:(snd_pcm_hw_delay) SNDRV_PCM_IOCTL_DELAY failed (-77): File descriptor in bad state when running mpv test.mkv --ao=alsa:device=iec958,alsa and pausing during playback. Historically, mplayer usually did not call snd_pcm_delay() (which is called by get_delay()) while paused, so this problem never showed up. But at least mpv has changes that cause get_delay() to be called when updating the status line (see commit 3f949cf). It's possible that calling snd_pcm_delay() is not always legal when the audio is paused, and at least fails with the error message mentioned above is the device is a hardware device. Change get_delay() to return the last delay before the audio was paused. The intention is to get a continuous playback status display, even when pausing or frame stepping, otherwise we could just return the audio buffer fill status in get_delay() or even just 0 when paused.
* ao_sdl: fix compilation with Libavwm42013-01-061-0/+1
| | | | | On Libav, <libavutil/fifo.h> doesn't recursively include common.h, but the code in ao_sdl.c uses some macros defined by this header.
* audio: make de-planarization fasterwm42012-12-283-16/+44
| | | | | | Uses the same trick as the planarization code to turn per-sample memcpy calls into mov instructions. Makes decoding a ~25min 48000Hz 2ch floatle audio file faster from 3.8s to 2.7s.
* sdl, encode_lavc: fix copyright headersRudolf Polzer2012-12-282-4/+4
| | | | | | Some of them had changes in 2012; extend their header. Fix project name.
* vo/ao: SDL 1.2+ audio driver, SDL 2.0+ accelerated video driverRudolf Polzer2012-12-282-0/+383
| | | | | | | | | | | This mainly serves as a fallback for platforms where nothing better is available; also as a debugging help. Both the audio and video driver are not first class - the audio driver lacks delay detection, and the video driver only supports a single YUV color space. Configure options: --disable-sdl2 to disable SDL 2.0+ detection, --disable-sdl to disable SDL 1.2+ detection. Both options need to be specified to turn off SDL support entirely.
* path: add mp_find_config_file and reorganize some of the codeStefano Pigozzi2012-12-151-6/+7
| | | | | | | | | | | | | | Add `mp_find_config_file` to search different known paths and use that in ass_mp to look for the fontconfig configuration file. Some incidental changes spawned by this feature where: * Buffer allocation for the strings containing the paths is now performed with talloc. All of the allocations are done on a NULL context, but it still improves readability of the code. * Move the OSX function for lookup inside of a bundle: this code path was currently not used by the bundle generated with `make osxbundle`. The plan is to use it again in a future commit to get a fontconfig config file.
* ao_lavc: stop using av_get_alt_sample_fmtRudolf Polzer2012-12-131-6/+6
| | | | Use av_get_planar_sample_fmt instead.
* cleanup: remove ao.brokenptswm42012-12-121-1/+0
| | | | This field was used by ao_v4l2, and is now unused.
* audio: remove support for native alaw/mulaw/adpcm outputwm42012-12-119-1228/+4
| | | | | | This is considered a worthless feature. Note that alaw/mulaw/adpcm input is unaffected: such data is handed to libavcodec and "decoded" to linear PCM.
* audio/decode: remove ad_dvdpcm and use ad_lavc for DVD PCMwm42012-12-112-164/+0
| | | | | | | | | | ad_dvdpcm reads MPEG specific headers directly (passed through codecdata by demux_mpg), so you couldn't use ffmpeg's "pcm_dvd" with demux_mpg. Change demux_mpg to set the correct audio parameters directly. The code for this is taken from ad_dvdpcm. ad_dvdpcm is evil because it still does partial packet reads (with demux_read_data()), and it's redundant to libavcodec anyway.
* audio/decode: remove ad_pcm and use ad_lavc for PCMwm42012-12-113-226/+95
| | | | | | | | | | | | | | | Since libavcodec doesn't have a "generic" PCM decoder, we have to go out of out way to make it look like ad_lavc provides one: make it provide a pseudo "pcm" decoder, which maps some format tags manually to the individual libavcodec PCM decoders. Format tags which uniquely map to one libavcodec could be mapped via codecs.conf. Since defining these in tag_map[] is much shorter (one line vs. a full codec entry in codecs.conf), and since we need tag_map[] anyway, we don't use codecs.conf for these. ad_pcm is evil because it still does partial packet reads (with demux_read_data()), and it's redundant to libavcodec anyway.
* audio: make AC3 pass-through with ad_spdif workreimar2012-12-034-24/+30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Do not fall back to 0 for samplerate when parser is not initialized. Might fix some issues with using -ac spdifenc with audio in MKV or MP4. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35517 b3059339-0415-0410-9bf9-f77b7e298cf2 Replace outdated list of unsupported formats by list of supported formats. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35534 b3059339-0415-0410-9bf9-f77b7e298cf2 Do not call af_fmt2str on the same data over and over. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35535 b3059339-0415-0410-9bf9-f77b7e298cf2 ad_spdif: use the more specific AF_FORMAT_AC3_LE when we handle AC3. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35536 b3059339-0415-0410-9bf9-f77b7e298cf2 Make AF_FORMAT_IS_IEC61937 include AF_FORMAT_IS_AC3. Our AC3 "sample format" is also iec61937. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35537 b3059339-0415-0410-9bf9-f77b7e298cf2 af_format: support endianness conversion also for iec61937 formats in general, not just AC3. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35538 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: audio/filter/af_format.c af_format: Fix check_format, non-special formats are of course supported. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35545 b3059339-0415-0410-9bf9-f77b7e298cf2 Note: see mplayer bug #2110
* audio: improve decoder open failure handlingUoti Urpala2012-12-031-0/+2
| | | | | | | | | | | | | | | | Reinitialize sh_audio->samplesize and sample_format before falling back to another audio decoder (some decoders rely on default values). Remove code setting these fields from demux_mkv and demux_lavf (no decoder should depend on demuxer-set values for these fields). Conflicts: audio/decode/ad_lavc.c Merged from mplayer2 commit 6b9567. The changes to ad_lavc.c are not merged, as they are very specific to the mplayer2 libavresample hack; we deplanarize manually, so we can't get unsupported sample formats yet (except on raw audio with "pcm_f64le", as we don't support AV_SAMPLE_FMT_DBL in the audio chain).
* ao_pcm: fix references to -novideowm42012-12-031-8/+1
| | | | | | | | | | | The option is -no-video. Remove the deprecated "fast" suboption, which did nothing and instructed the user to use "-novideo" instead. Fix a reference to -novideo in encoding.rst. Add a "generic" entry about -no-* to the list of renamed options. The change is already explicitly mentioned in the text above the table, but even if it's redundant, it makes it harder to overlook.
* af_lavcac3enc, encode: support planar formatsRudolf Polzer2012-12-034-7/+138
| | | | | | | This fixes operation with current ffmpeg releases. Note that this planarization is slow and should be reverted once proper planar audio support is there in mpv.
* ao_pulse: do not allow setting volume over 100%wm42012-11-241-2/+2
| | | | | | | | | | | | | PulseAudio allows applications to set volume over 100%. To make this possible, the PulseAudio daemon raises the global system volume, and tries to lower other applications volumes. Unfortunately, this doesn't work out and doesn't manage to keep the effective volume level of these other applications. To make it short: this functionality invoked PulseAudio bugs. Disable it. This essentially reverts commit 85a64b.
* core: fix crash when video filter returns inf as PTSwm42012-11-201-1/+1
| | | | | | | | | | | | | | | | | | | When a video filter returned inf as PTS, the player crashed. One reason for this was that decode_audio() was called with a negative minlen parameter, which at some point caused it to call a memory allocation function with a ridiculous value, triggering an out of memory code path in talloc.c. (talloc.c has been modified to abort() on out of memory situations.) Fix this by sanity checking minlen in decode_audio(). (The check against outbuf->len always succeeded, because it's an unsigned comparison.) Make an existing sanity check in mplayer.c more robust: check for NaN too, which happens if the video PTS is inf. This happened with "-vf pullup,softpulldown" (but is not triggered when the following commit is applied).
* Fix potential bugs and issues, general cleanupsreimar2012-11-203-5/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Most of these are reimar fixing issues found by Coverity static analyzer, and possibly some more cleanup commits independent from this. Since these commits are rather noisy, squash them all together. Try to make code a bit clearer. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35294 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: audio/out/ao_alsa.c Check the correct variable for NULL. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35323 b3059339-0415-0410-9bf9-f77b7e298cf2 Remove pointless unreachable code (the loop condition already checks the 0xff case). git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35325 b3059339-0415-0410-9bf9-f77b7e298cf2 Fix typo that might have caused reading beyond the string end. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35326 b3059339-0415-0410-9bf9-f77b7e298cf2 Do not needlessly use "long" types. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35331 b3059339-0415-0410-9bf9-f77b7e298cf2 Use AV_RB32 to avoid sign extension issues and validate offset before using it. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35332 b3059339-0415-0410-9bf9-f77b7e298cf2 Remove nonsense casts. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35343 b3059339-0415-0410-9bf9-f77b7e298cf2 Fix crash in case sh_audio allocation failed. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35348 b3059339-0415-0410-9bf9-f77b7e298cf2 Fix potential NULL dereference. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35351 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: libmpcodecs/ad_ffmpeg.c Note: Slightly modified. Fix malloc failure check to check the correct variable. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35353 b3059339-0415-0410-9bf9-f77b7e298cf2 Avoid code duplication and pointless casts. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35363 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: stream/tv.c Error out if an invalid channel list name was specified instead of continuing and reading outside array bounds all over the place. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35364 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: stream/tv.c Make array "static const". git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35365 b3059339-0415-0410-9bf9-f77b7e298cf2 Properly free resources even when encountering many parse errors. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35367 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: parser-cfg.c Avoid leaks in error handling. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35380 b3059339-0415-0410-9bf9-f77b7e298cf2 Do not do sign comparisons on "char" type which can be both signed or unsigned. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35381 b3059339-0415-0410-9bf9-f77b7e298cf2 Free cookies file data after parsing it. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35382 b3059339-0415-0410-9bf9-f77b7e298cf2 http_set_field only makes a copy of the string, so we still need to free it. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35383 b3059339-0415-0410-9bf9-f77b7e298cf2 check4proxies does not modify input URL, so mark it const. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35390 b3059339-0415-0410-9bf9-f77b7e298cf2 Remove proxy "support" from stream_rtp and stream_upd, trying to use a http proxy for UDP connections makes no sense. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35394 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: stream/stream_rtp.c stream/stream_udp.c Add url_new_with_proxy function to reduce code duplication and memleaks. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35395 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: stream/pnm.c stream/stream_live555.c stream/stream_nemesi.c stream/stream_rtsp.c Fix off-by-one errors in file descriptor validity checks. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35402 b3059339-0415-0410-9bf9-f77b7e298cf2 Remove pointless cast. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35403 b3059339-0415-0410-9bf9-f77b7e298cf2 Abort when opening the file failed instead of calling "write" with an invalid descriptor. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35404 b3059339-0415-0410-9bf9-f77b7e298cf2 Remove pointless local variable. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@35411 b3059339-0415-0410-9bf9-f77b7e298cf2 Conflicts: stream/http.c
* Improve compatibility with Libav 0.8.4 and ffmpeg 0.11.2wm42012-11-141-0/+1
| | | | | | | | | | | Libav 0.8.4 is ridiculously old (in relative terms), so I don't know how many things are broken silently. Encoding is disabled, because the required API hasn't been added yet. (On the other hand, the old API can't be used in newer versions.) This should improve compatibility with ffmpeg 0.11.2 as well, which didn't define AV_CODEC_ID_SUBRIP yet.
* mixer: fix lowering hw volume while mutedUoti Urpala2012-11-141-1/+1
| | | | | | | Lowering volume while muted did not work correctly with audio outputs that support native mute setting separate from volume (ao_alsa and ao_pulse), because the AO-level volume was not set while muted but was still being read back. Fix by setting the AO volume in this case.
* ao_coreaudio: fix deprecation warningsStefano Pigozzi2012-11-131-6/+6
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* Add missing compat/libav.h includeswm42012-11-121-0/+1
| | | | For avcodec_free_frame().
* Rename directories, move files (step 2 of 2)wm42012-11-1235-98/+93
| | | | | | | | | | | | Finish renaming directories and moving files. Adjust all include statements to make the previous commit compile. The two commits are separate, because git is bad at tracking renames and content changes at the same time. Also take this as an opportunity to remove the separation between "common" and "mplayer" sources in the Makefile. ("common" used to be shared between mplayer and mencoder.)
* Rename directories, move files (step 1 of 2) (does not compile)wm42012-11-1268-0/+22057
Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code.