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* ao_opensles: remove 32bit audioGravatar Josh de Kock2016-05-22
| | | | It's unsupported by android, and can cause problems when trying to play 32bit audio. Removing 32bit fixes it by forcing 16 bit or 8 bit audio.
* ao_alsa: add more shitty workaroundsGravatar wm42016-05-06
| | | | | | | | | | | | | | | This reportedly makes it work on ODROID-C2. The idea for this hack is taken from kodi; they unconditionally set some or all of those flags. I don't trust ALSA enough to hope that setting these flags couldn't break something else, so we try without them first. It's not clear whether this is a driver bug or a bug in the ALSA libs. There is no ALSA bug tracker (the ALSA website has had a dead link to a deleted bug tracker fo years). There's not much we can do other than piling up ridiculous hacks. At least I think that at this point invalid API usage by mpv can be excluded as a cause. ALSA might be the worst audio API ever.
* ao_alsa: log final hwparams tooGravatar wm42016-05-03
| | | | snd_pcm_hw_params() updates them.
* win32: replace libuuid.a usage with initguid.hGravatar James Ross-Gowan2016-05-01
| | | | | | | | | | | | | | | Including initguid.h at the top of a file that uses references to GUIDs causes the GUIDs to be declared globally with __declspec(selectany). The 'selectany' attribute tells the linker to consolidate multiple definitions of each GUID, which would be great except that, in Cygwin and MinGW GCC 6.1, this method of linking makes the GUIDs conflict with the ones declared in libuuid.a. Since initguid.h obsoletes libuuid.a in modern compilers that support __declspec(selectany), add initguid.h to all files that use GUIDs and remove libuuid.a from the build. Fixes #3097
* ao_alsa: log hwparams while restricting themGravatar wm42016-04-28
| | | | | | They can sometimes fail, so I want logging to determine what's going on. Most of them are at debug log-level, except the final hwparams.
* ao_coreaudio: remove detected_deviceGravatar wm42016-04-26
| | | | | | | | | | | Setting this here is a race condition. It's called from a CoreAudio callbacks, and there are no locks. It's a string, so this can be potentially severe. It's hard to fix and only CoreAudio supported it, so remove it. This causes the "audio-out-detected-device" property to return nothing on all platforms.
* ad_spdif: take care of deprecated libavcodec API usageGravatar wm42016-04-20
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* ao_coreaudio_exclusive: list formats when searching substreamGravatar wm42016-04-15
| | | | Should help debug problems with AC3 passthrough not working.
* ao_coreaudio: remove unused functionGravatar wm42016-04-15
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* encode_lavc: Migrate to codecpar API.Gravatar Rudolf Polzer2016-04-11
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* ao_coreaudio_exclusive: add missing newline to log messageGravatar wm42016-04-01
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* demux_lavf, ad_lavc, ad_spdif, vd_lavc: handle FFmpeg codecpar API changeGravatar wm42016-03-31
| | | | | | | | | AVFormatContext.codec is deprecated now, and you're supposed to use AVFormatContext.codecpar instead. Handle this for all of the normal playback code. Encoding mode isn't touched.
* ad_lavc, vd_lavc: support new Libav decoding APIGravatar wm42016-03-24
| | | | For now only found in Libav.
* ad_lavc: add codec_timebase hack tooGravatar wm42016-03-24
| | | | | vd_lavc.c had this, and soon I'll need it in ad_lavc.c too. For now it's unused.
* ao_lavc: use new af_select_best_samplerate functionGravatar Kevin Mitchell2016-03-17
| | | | | | | | This is particularly useful for opus which allows only a fairly restrictive set of samplerates. If the codec doesn't provide a list of samplerates, just continue to try the requsted one and hope for the best. fixes #2957
* ao_wasapi: use new af_select_best_samplerate functionGravatar Kevin Mitchell2016-03-17
| | | | It duplicates the logic that was previously used here.
* audio: add af_select_best_samplerate functionGravatar Kevin Mitchell2016-03-17
| | | | | | This function chooses the best match to a given samplerate from a provided list. This can be used, for example, by the ao to decide what samplerate to use for output.
* ao_wasapi: make wait for audio thread termination infiniteGravatar Kevin Mitchell2016-02-26
| | | | | The time-out was a terrible hack for marginally better behaviour when encountering #1773, which appears to have been resolved by a previous commit.
* ao_wasapi: further flatten/simplify volume controlGravatar Kevin Mitchell2016-02-26
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* ao_wasapi: use MP_FATAL for stuff that leads to init failureGravatar Kevin Mitchell2016-02-26
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* ao_wasapi: move pre-resume reset into resume functionGravatar Kevin Mitchell2016-02-26
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* ao_wasapi: move resetting the thread state into main loopGravatar Kevin Mitchell2016-02-26
| | | | | This was previously duplicated between the reset/resume functions, and not properly handled in the "impossible" invalid thread state case.
* ao_wasapi: set buffer size to device period in exclusive modeGravatar Kevin Mitchell2016-02-26
| | | | | | | | | | | | | This eliminates some intermittent pops heard in a HRT MicroStreamer DAC uncorrelated with user interaction. As a bonus, this resolves #1773 which I can o longer reproduce as of this commit. Leave the 50ms buffer for shared mode since that seems to be working quite well. This is also the way exclusive mode is done in the MSDN example code: https://msdn.microsoft.com/en-us/library/windows/desktop/dd370844%28v=vs.85%29.aspx This was originally increased in c545c40 to mitigate glitches that subsequent refactorings have eliminated.
* ao_wasapi: replace laggy COM messaging with mp_dispatch_queueGravatar Kevin Mitchell2016-02-26
| | | | | | | | | | A COM message loop is apparently totally inappropriate for a low latency thread. It leads to audio glitches because the thread doesn't wake up fast enough when it should. It also causes mysterious correlations between the vo and ao thread (i.e., toggling fullscreen delays audio feed events). Instead use an mp_dispatch_queue to set/get volume/mute/session display name from the audio thread. This has the added benefit of obviating the need to marshal the associated interfaces from the audio thread.
* ao_wasapi: avoid under-run cascade in exclusive mode.Gravatar Kevin Mitchell2016-02-26
| | | | | | | | | | | | | | | | Don't wait for WASAPI to send another feed event if we detect an underfull buffer. It seems that WASAPI doesn't always send extra feed events if something causes rendering to fall behind. This causes every subsequent playback buffer to under-run until playback is reset. The fix is simply to do a one-shot double feed when this happens, which allows rendering to catch up with playback. This was observed to happen when using MsgWaitForMultipleObjects to wait for the feed event and toggling fullscreen with vo=opengl:backend=win. This commit improves the behaviour in that specific case and more generally makes exclusive mode significantly more robust. This commit also moves the logic to avoid *over*filling the exclusive mode buffer into thread_feed right next to the above described underfil logic.
* ao_wasapi: fix typo in commentGravatar Kevin Mitchell2016-02-26
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* ao_wasapi: use SUCCEEDED/FAILED macrosGravatar Kevin Mitchell2016-02-26
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* ao: initial OpenSL ES supportGravatar Ilya Zhuravlev2016-02-27
| | | | | | | | OpenSL ES is used on Android. At the moment only stereo output is supported. Two options are supported: 'frames-per-buffer' and 'sample-rate'. To get better latency the user of libmpv should pass values obtained from AudioManager.getProperty(PROPERTY_OUTPUT_FRAMES_PER_BUFFER) and AudioManager.getProperty(PROPERTY_OUTPUT_SAMPLE_RATE).
* audio: make mp_audio_skip_samples() adjust the PTSGravatar wm42016-02-22
| | | | Slight simplification/cleanup.
* ad_lavc: skip AVCodecContext.delay samples at beginningGravatar wm42016-02-22
| | | | | | | | Fixes correctness_trimming_nobeeps.opus. One nasty thing is that this mechanism interferes with the container-signalled mechanism with AV_FRAME_DATA_SKIP_SAMPLES. So apply it only if that is apparently not present. It's a mess, and it's still broken in FFmpeg CLI, so I'm sure this will get fucked up later again.
* ad_lavc: make sample trimming symmetric to skippingGravatar wm42016-02-22
| | | | | | I'm not quite sure what the FFmpeg AV_FRAME_DATA_SKIP_SAMPLES API demands here. The code so far assumed that skipping can be more than a frame, but not trimming. Extend it to trimming too.
* ad_lavc: move skipping logic out of the HAVE_AVFRAME_SKIP_SAMPLES blockGravatar wm42016-02-22
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* ad_lavc: interpolate missing timestampsGravatar wm42016-02-22
| | | | | | | | | | This is actually already done by dec_audio.c. But if AV_FRAME_DATA_SKIP_SAMPLES is applied, this happens too late here. The problem is that this will slice off samples, and make it impossible for later code to reconstruct the timestamp properly. Missing timestamps can still happen with some demuxers, e.g. demux_mkv.c with Opus tracks. (Although libavformat interpolates these itself.)
* audio: move frame clipping to a generic functionGravatar wm42016-02-21
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* Rewrite ordered chapters and timeline stuffGravatar wm42016-02-15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This uses a different method to piece segments together. The old approach basically changes to a new file (with a new start offset) any time a segment ends. This meant waiting for audio/video end on segment end, and then changing to the new segment all at once. It had a very weird impact on the playback core, and some things (like truly gapless segment transitions, or frame backstepping) just didn't work. The new approach adds the demux_timeline pseudo-demuxer, which presents an uniform packet stream from the many segments. This is pretty similar to how ordered chapters are implemented everywhere else. It also reminds of the FFmpeg concat pseudo-demuxer. The "pure" version of this approach doesn't work though. Segments can actually have different codec configurations (different extradata), and subtitles are most likely broken too. (Subtitles have multiple corner cases which break the pure stream-concatenation approach completely.) To counter this, we do two things: - Reinit the decoder with each segment. We go as far as allowing concatenating files with completely different codecs for the sake of EDL (which also uses the timeline infrastructure). A "lighter" approach would try to make use of decoder mechanism to update e.g. the extradata, but that seems fragile. - Clip decoded data to segment boundaries. This is equivalent to normal playback core mechanisms like hr-seek, but now the playback core doesn't need to care about these things. These two mechanisms are equivalent to what happened in the old implementation, except they don't happen in the playback core anymore. In other words, the playback core is completely relieved from timeline implementation details. (Which honestly is exactly what I'm trying to do here. I don't think ordered chapter behavior deserves improvement, even if it's bad - but I want to get it out from the playback core.) There is code duplication between audio and video decoder common code. This is awful and could be shareable - but this will happen later. Note that the audio path has some code to clip audio frames for the purpose of codec preroll/gapless handling, but it's not shared as sharing it would cause more pain than it would help.
* audio/video: expose codec info as separate fieldGravatar wm42016-02-15
| | | | | Preparation for the timeline rewrite. The codec will be able to change, the stream header not.
* ad_lavc: fix --ad-lavc-threads rangeGravatar wm42016-02-11
| | | | | | | The code is shared with the --vd-lavc-threads option, so using 0 for auto-detection just works. But no, this is not useful. Just change it for orthogonality.
* Initial Android supportGravatar Jan Ekström2016-02-10
| | | | | * Adds an 'android' feature, which is automatically detected. * Android has a broken strnlen, so a wrapper is added from FreeBSD.
* audio: minor simplificationGravatar wm42016-02-05
| | | | | These fields are already deallocated by uninit_decoder(). Also remove the wrong/useless log message.
* build: make libavfilter mandatoryGravatar wm42016-02-05
| | | | | | The complex filter support that will be added makes much more complex use of libavfilter, and I'm not going to bother with adding hacks to keep libavfilter optional.
* ao_coreaudio: fix 7.1(rear) channel mappingGravatar wm42016-02-04
| | | | | | | | | | | | | | I can't explain this, but it seems to be a similar case to the ALSA HDMI one. I find it hard to tell because of the slightly different names and conventions in use in libavcodec, WAVEEXT channel masks, decoders, codec specifications, HDMI, and platform audio APIs. The fix is the same as the one for ao_alsa (see commit be49da72). This should fix at least playing 7.1 sources on OSX with 7.1(rear) selected in Audio MIDI Setup. The ao_alsa commit mentions XBMC, but I couldn't find out where it does that or if it also does that for CoreAudio. It's woth noting that PHT (essentially an old XBMC fork) also exhibited the incorrect behavior (i.e. side and back speakers were swapped).
* af_lavrresample: change fudged channelsGravatar wm42016-02-04
| | | | | | | | | | | | | Remove flc-frc <-> sl<->sr. This was just plain wrong, and a mistaken change to make 7.1 work properly on CoreAudio with 7.1(rear) layout. Also see the following commit. Add br-br <-> sl<->sr, because we decided that it makes sense. Note that this "fudging" is applied only if the channel pairs are replaced, i.e. they would get dropped and be replaced with silence. This is done to compensate for libswresample's default rematrixing (which takes care of some more common cases).
* audio/video: merge decoder return valuesGravatar wm42016-02-01
| | | | | | Will be helpful for the coming filter support. I planned on merging audio/video decoding, but this will have to wait a bit longer, so only remove the duplicate status codes.
* Fix build on LibavGravatar wm42016-01-30
| | | | I hope.
* audio: move pts reset checkGravatar wm42016-01-29
| | | | Reduces the dependency of the filter/output code on the decoder.
* audio: move mp_audio->AVFrame conversion to a functionGravatar wm42016-01-29
| | | | | | | | | This also makes it refcounted, i.e. the new AVFrame will reference the mp_audio buffers, instead of potentially forcing the consumer of the AVFrame to copy the data. All the extra code is for handling the >8 channels case, which requires very messy dealing with the extended_ fields (not our fault).
* ao_wasapi: add "wasapi" prefix to non-static find_deviceID functionGravatar Kevin Mitchell2016-01-28
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* ao_wasapi: correct check for specified device on default changeGravatar Kevin Mitchell2016-01-28
| | | | | | Correctly avoid a reload if the current device was specified by the user through --audio-device. Previously, we only recognized if the user had specified --ao=wasapi:device=.
* ao_wasapi: fix check for already found deviceGravatar Kevin Mitchell2016-01-28
| | | | | oops, forgot to change this when I made get_deviceID a more proper function. state->deviceID is not set or read here - that's for the caller to do.
* command: always allow setting volume/mute propertiesGravatar wm42016-01-26
| | | | | | | | | | | | | | | | | | | | | | | This seems generally easier when using libmpv (and was already requested and implemented before: see commit 327a779a; it was reverted some time later). With the weird internal logic we have to deal with, in particular the --softvol=no case (using system volume), and using the audio API's mixer (--softvol=auto on some systems), we still can't avoid all glitches and corner cases that complicate this issue so much. The API user is either recommended to use --softvol=yes or auto, or to watch the new mixer-active property, and assume the volume/mute properties have significant values if the mixer is active. Remaining glitches: - changing the volume/mute properties has no effect if no internal mixer is used (--softvol=no) and the mixer is not active; the actual mixer controls do not change, only the property values - --volume/--mute do not have an effect on the volume/mute properties before mixer initialization (the options strictly are only applied during mixer init) - volume-max is 100 while the mixer is not active
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