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* af: fix behavior with pathologic filter chainswm42015-07-071-0/+2
| | | | | | | | Some filter chains require a huge number of auto-inserted conversion filters. There is an overly stupid safeguard against infinite filter insertions, which counts the number of conversion filters inserted. This triggered accidentally in this case. Fix by resetting this counter after a non-conversion filter was successfully configured.
* ao_coreaudio: fix device latency, share the codewm42015-07-064-23/+25
| | | | | | | | | ao_coreaudio (using AudioUnit) accounted only for part of the latency - move the code in ao_coreaudio_exclusive to utils, and use that for the AudioUnit code. (There's still the question why CoreAudio and AudioUnit require you to jump through hoops this much, but apparently that's how it is.)
* ao_coreaudio_exclusive: continue even if setting physical format failswm42015-07-062-2/+4
| | | | | Makes it work with (apparently) crappy drivers, which refuse to set the physical format in some cases.
* ao_coreaudio_exclusive: fix some verbose outputwm42015-07-041-2/+2
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* ao: don't pass along AO arguments when redirectingwm42015-07-031-1/+1
| | | | Only causes problems.
* ao_coreaudio: add exclusive suboptionwm42015-07-031-2/+4
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* ao_coreaudio_exclusive: support PCMwm42015-07-031-15/+35
| | | | | | | | | | | | | | | | | | Until now, this was for AC3 only. For PCM, we used AudioUnit in ao_coreaudio, and the only reason ao_coreaudio_exclusive exists is that there is no other way to passthrough AC3. PCM support is actually rather simple. The most complicated issue is that modern OS X versions actually do not support copying through the data; instead everything must go through float. So we have to deal with virtual and physical format being different, which causes some complications. This possibly also doesn't support some other things correctly. For one, if the device allows non-interleaved output only, we will probably fail. (I couldn't test it, so I don't even know what is required. Supporting it would probably be rather simple, and we already do it with AudioUnit.)
* ao_coraudio: reject all non-PCM formatswm42015-07-031-1/+1
| | | | | Currently this is equivalent. On the other hand, all audio code should reject formats that is not in a category known to it.
* ao_coreaudio_utils: reduce spamwm42015-07-031-1/+0
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* ao_coreaudio_utils: fix format back-mappingwm42015-07-031-5/+7
| | | | | | | | | | | Mapping of spdif formats was imperfect. Since the first format on the list is somehow AAC, it was returned first, which is confusing, because CoreAudio calls all spdif formats AC3. Since the spdif formats have some rather arbitrary, reverse mapping the formats didn"t actually work either. Fix by explicitly ignoring these when spdif is used. Also, don't forget to set the samplerate in ca_asbd_to_mpformat(), or it will work only in some cases.
* ao_coreaudio_exclusive: do not set ao->bpswm42015-07-031-3/+0
| | | | | This field is basically deprecated or for convenience only, and this code doesn't need it.
* ao_coreaudio_exclusive: dump all latency info in verbose modewm42015-07-031-1/+4
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* ao_coreaudio_exclusive: factor format selectionwm42015-07-031-35/+44
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* ao_coreaudio_exclusive: separate out stream selectionwm42015-07-031-61/+76
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* af_lavrresample: log actual channel layout conversionswm42015-06-301-3/+6
| | | | | With all the reordering etc. that can go on in this filter, it's useful to see what upmix/downmix it's actually performing.
* ao_coreaudio: restore physical format if it can't be set exactlywm42015-06-302-2/+14
| | | | | | | | May help with (supposedly) bad drivers, which can put the device into some sort of broken state when trying to set a different physical format. When the previous format is restored, it apparently recovers. This might make the change-physical-format suboption more robust.
* ao_coreaudio: support native mono outputwm42015-06-291-0/+2
| | | | | | | We can be pretty sure that AudioUnit will remix for us. Before this commit, we usually upmixed to stereo, because the stereo and multichannel layouts were the only whitelisted ones.
* ao_coreaudio: log hotplug events explicitlywm42015-06-291-3/+6
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* ao_wasapi: fix regressionwm42015-06-271-1/+1
| | | | This probably fixes the regression introduced with commit 6147bcce.
* chmap_sel: add a mechanism for preferred conversionswm42015-06-271-9/+39
| | | | | And use it for the mono case. This is slightly more formal and will make it easier to add more such cases.
* chmap_sel: remove outdated incomplete commentwm42015-06-271-3/+0
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* audio: fix format function consistency issueswm42015-06-2620-86/+83
| | | | | | | | | | | Replace all the check macros with function calls. Give them all the same case and naming schema. Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes(). Introduce af_fmt_is_pcm(), and use it in situations that used !AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format was. It simply meant "not PCM".
* audio: replace format name tablewm42015-06-263-52/+23
| | | | Having a big switch() is simpler.
* audio: remove bitmask format definition messwm42015-06-262-92/+80
| | | | | | | | | | Audio formats used a semi-clever schema to encode the properties of the PCM encoding as bitfields into the format integer value. The af_fmt_change_bits() implementation becomes a bit weird, but it's an improvement to the rest of the code. (I've always disliked it, so why not get rid of it.)
* ao_coreaudio_utils: use a macrowm42015-06-261-1/+1
| | | | | This is actually the last line of code outside of format.c/h which still tries to fiddle with the format bitfields.
* ao_sndio: fix commentwm42015-06-261-2/+1
| | | | | | So whoever (nobody?) would want to deal with this broken and obscure AO for an obscure audio API could add support for some more channel layouts.
* ao_coreaudio: support non-interleaved outputwm42015-06-262-5/+12
| | | | | This saves us the trouble of interleaving the audio data for no reason.
* ao_coreaudio: explicitly skip input streamswm42015-06-262-4/+22
| | | | | | | This may or may not fix some issues with the format switching code. Actually, it seems somewhat unlikely, but then checking the stream type isn't incorrect either, and is probably something the API user should always be doing.
* ao_coreaudio_utils: compare full AudioStreamBasicDescriptionwm42015-06-251-1/+3
| | | | | | Originally, this was written for comparing the sample format only, but ca_change_physical_format_sync() actually expects that the full format is compared. (For all other uses it doesn't matter.)
* audio: output human-readable channel layouts toowm42015-06-254-10/+30
| | | | | This gets you the "logical" channel layout, instead of the exact thing we're sending to the AO. (Tired of the cryptic shit ALSA gives me.)
* audio: fix channel map fallback selection (again)wm42015-06-251-36/+61
| | | | | | | | | | | | | | | | | | | | The speaker replacement nonsense sometimes made blatantly incorrect decisions. In this case, it prefered a 7.1(rear) upmix over outputting 5.1(side) as 5.1, which makes no sense at all. This happened because 5.1 and 7.1(rear) appeared equivalent to the final selection, as both of them lose the sl-sr channels. The old code was too stupid to select the one with the lower number of channels as well. Redo this. There's really no reason why there should be a separate final decision, so move the speaker replacement logic into the mp_chmap_is_better() function. Improve some other details. For example, we never should compare the plain number of channels for deciding upmix/downmix, because due to NA channels this is essentially meaningless. Remove the NA channels when doing this comparison. Also, explicitly handle exact matches. Conceptually this is not necessary, but it avoids that we have to needlessly shuffle audio data around.
* ao: standardize channel layout name in debug output furtherwm42015-06-251-0/+2
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* af: move af_from_dB() function to af_volume.cwm42015-06-233-22/+11
| | | | | And also simplify it (it certainly had the most awkward API you could think of for such a simple function).
* af_volume: add a replaygain fallback optionwm42015-06-231-0/+5
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* af_lavrresample: free and reallocate resample context on reconfigwm42015-06-221-30/+35
| | | | | | | | This avoids keeping "bad" state from previous reconfig calls, such as the internal_sample_format option (which is set only on the first reconfig call). There's no advantage to keeping the resample contexts around anyway.
* af_lavrresample: fix commentwm42015-06-221-4/+4
| | | | | mp_format is not a libavresample input format here, and the comment was more confusing than it helped.
* af: restore detaching of PCM filters when using spdifwm42015-06-223-1/+14
| | | | | | Basically, af_fix_format_conversion() behaves stupid you insert a conversion filter that won't work, and adding back the conversion test function is the simplest fix to it.
* af_lavrresample: don't flush in uninitialized statewm42015-06-221-1/+2
| | | | libswresample verbosely complains.
* ao_alsa: fix a log messagewm42015-06-211-4/+5
| | | | | | | | | | | | | | | | | So apparently, this essentially happens when the kernel driver doesn't implement write accesses in the channel map control. Which doesn't necessarily mean that the channel map is unsupported, or that there is a bug - it's just lazyness and a consequence of the terrible ALSA kernel API for the channel mapping stuff. In these cases, the channel count implicitly selects the channel map, and snd_pcm_set_chmap() always fails with ENXIO. I'm actually not sure what happens if dmix is on top of e.g. HDMI, which actually lets you change the channel mapping. I'm also not sure why commit d20e24e5d1614354e9c8195ed0b11fe089c489e4 (alsa-lib git repository) does not take care of this.
* demux: merge extradata fieldswm42015-06-211-5/+1
| | | | | | | MPlayer traditionally had completely separate sh_ structs for audio/video/subs, without a good way to share fields. This meant that fields shared across all these headers had to be duplicated. This commit deduplicates essentially the last remaining duplicated fields.
* demux: rename sh_stream.format to sh_stream.codec_tagwm42015-06-211-1/+1
| | | | | Why not. "format" sounds too misleading for the actual importance and meaning of this field.
* Various spelling fixesMarcin Kurczewski2015-06-188-13/+13
| | | | Signed-off-by: wm4 <wm4@nowhere>
* ao_wasapi: fix crash on hotplug init errorwm42015-06-171-0/+1
| | | | | On init error, the mp_msg macros are actually called. They could cause a crash because state->log was NULL.
* af_lavrresample: include osdep/endian.hwm42015-06-171-0/+1
| | | | The 24 bit conversion code needs the relevant preprocessor symbols.
* af: remove conversion filter searchwm42015-06-163-94/+4
| | | | | | This attempted to find a minimal filter graph for a format conversion involving multiple conversion filters. With the last 2 commits it becomes dead code - remove it.
* af_convert24: remove this filterwm42015-06-162-123/+0
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* af_lavrresample: integrate 24 bit (3 bytes per sample) outputwm42015-06-161-9/+48
| | | | | | | | | | | | Now af_lavrresample can output 24 bit samples directly, by doing the conversion "inline". Luckily, S32->S24 can be done in-place, so this isn't too much work. But the output conversion logic (which seems to be adding up) gets slightly more complicated again. Normally this is done by af_convert24. But having multiple conversion filters complicates some aspects of the filter chain. S24 output is the only thing the code for multiple conversion filters is still needed for, and getting rid of that is preferable.
* af_lavrresample: always fill reorderwm42015-06-161-1/+1
| | | | | | If the code path for additional output conversion is active, reorder_planes() is always called, even if the reorder_out array wasn't filled. This is obviously wrong - always fill this array.
* audio: remove S8, U16, U24, U32 formatswm42015-06-1612-197/+26
| | | | | | | | | | | | | They are useless. Not only are they actually rarely in use; but libavcodec doesn't even output them, as libavcodec has no such sample formats for decoded audio. Even if it should happen that we actually still need them (e.g. if doing direct hardware output), there are better solutions. Swapping the sign is a fast and lossless operation and can be done inplace, so AO actually needing it could do this directly. If you wonder why we keep U8 instead of S8: because libavcodec does it.
* audio: fix crash on uninitwm42015-06-151-1/+1
| | | | Shit.
* af_lavcac3enc: fix A/V syncwm42015-06-151-3/+17
| | | | | | The filter can buffer singificant amounts of audio. (The proper fix is making the filter chain PTS-aware.)
* af: fix an aspect of filter chain flushingwm42015-06-151-0/+11
| | | | | Even if we flush the current filter, we have to read the remaining output from the frame we previously fed to the filter.
* audio: remove unused readonly fieldwm42015-06-152-3/+1
| | | | Its last use was removed in 433402b5.
* chmap: make up some channel layout nameswm42015-06-121-0/+3
| | | | | Going by the existing names, these should make sense. HDMI knows about these layouts, but does not name them.
* ao_alsa: if possible, reorder device maps to std layoutswm42015-06-121-0/+2
| | | | | | | Channel maps reported by the device as SND_CHMAP_TYPE_VAR can be freely reordered. We don't use this much (out of laziness), but in this case it's a simple way to reduce necessary reordering (which would be an extra libavresample invocation), and to make debug output more readable.
* ao_alsa: make it accept 7.1 over HDMIwm42015-06-121-0/+2
| | | | | SDR/SDL is what lavc outputs for 7.1(rear), while RRC/RLC is what ALSA uses for some 7.1 layouts, so this makes sense to me.
* chmap_sel: improve speaker replacement handlingwm42015-06-121-17/+33
| | | | | This didn't really work since the last time the channel map fallback code was touched. In some cases, quite bad results were selected.
* chmap_sel: do naive speaker replacements lastwm42015-06-121-3/+4
| | | | | This prevents that the potentially better pick by mp_chmap_sel_fallback() is overridden.
* audio: fill NA channels with silencewm42015-06-121-11/+29
| | | | | | | | | | | | | | | Until now, we didn't do this, because it required some effort, and didn't seem to be necessary. It probably still isn't, but it sounds like a good idea not to output arbitrary data on these channels. The situation is complicated by the fact that just adding new channels to a planar frame would require messing with buffers. So we would have to allocate new buffers and add them to the frame. We could have to maintain an extra buffer pool for this. Avoid this by being "clever", and just allocate a frame with enough channels in the first place. libav/swresample won't know about these channels and won't write to them, but we can grab them in reorder_planes() and use them for the NA channels.
* audio: use unknown channel layouts if there is no standard layoutwm42015-06-121-5/+5
| | | | | | | | | This is just a conceptual issue, since for now every channel count has an associated standard layout. But should the max. channel count ever be bumped, some things would stop function if mp_chmap_from_channels() refused to work for any channel count within the allowed range.
* audio: fix messed up channel reorderingwm42015-06-121-2/+2
| | | | Quite a blunder, really.
* audio: deal with AVFrame-style buffer assignmentswm42015-06-122-4/+17
| | | | | | | | | | | | | | | | In the AVFrame-style system (which we inreasingly map our internal data stuctures on), buffers and plane pointers don't necessarily have a 1:1 correspondence. For example, a single buffer could cover 2 or more planes, all while other planes are covered by a second buffer, and so on. They don't need to be ordered in the same way. Change mp_audio_get_allocated_size() to retrieve the maximum size all planes provide. This also considers the case of planes not pointing to buffer start. Change mp_audio_realloc() to reset all planes, even if corresponding buffers are not reallocated. (The caller has to be careful anyway if it wants to be sure the contents are preserved on realloc calls.)
* ao_alsa: change ALSA braindeath heuristicwm42015-06-111-3/+4
| | | | | | | | | | | | If you try to play surround with dmix, it will advertise surround and lets you set more than 2 channels, but will report a stereo channel map, with the extra channels identified as NA. We could handle this now, but we don't want to (because it's excessively stupid). Do it only if the channel map is not what we requested, instead of just acting if it contains NA entries at all. This avoids that we hurt ourselves in the unlikely but possible case we actually have to use channel maps with NA entries.
* ao_coreaudio: change physical stream format synchronouslywm42015-06-092-5/+2
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* audio/out/pull: avoid dropping some audio when drainingwm42015-06-091-9/+18
| | | | | | | | | | | | If the audio API takes a while for starting the audio callback, the current heuristic can be off. In particular, with very short files, it can happen that the audio callback is not called before playback is stopped, so no audio is output at all. Change draining so that it essentially waits for the ringbuffer to empty. The assumption is that once the audio API has read the data via the callback, it will always output it, even if the audio API is stopped right after the callback has returned.
* audio/out/pull: correctly pad partial frames with silencewm42015-06-091-1/+1
| | | | | | | | If a frame could only be partially filled with real audio data, the silence wasn't written at the correct offset. It could have happened that the remainder of the frame contained garbage. (This didn't happen in the more common case of playing dummy silence.)
* ao_alsa: refine channel count mismatch error messagewm42015-06-091-2/+2
| | | | I suspect we need to hand this more gracefully in some cases.
* audio: add --audio-spdif as new method for enabling passthroughwm42015-06-052-8/+21
| | | | | | | | | | | | | This provides a new method for enabling spdif passthrough. The old method via --ad (--ad=spdif:ac3 etc.) is deprecated. The deprecated method will probably stop working at some point. This also supports PCM fallback. One caveat is that it will lose at least 1 audio packet in doing so. (I don't care enough to prevent this.) (This is named after the old S/PDIF connector, because it uses the same underlying technology as far as the higher level protoco is concerned. Also, the user should be renamed that passthrough is backwards.)
* ad_spdif: use a pseudo codec entry to select DTS-HD instead of an optionwm42015-06-051-1/+10
| | | | | | | | | This deprecates the --ad-spdif-dtshd option, and replaces it with a pseudo decoder. This means ad_spdif will report two decoders, "dts" and "dts-hd", of which the second simply enables what the option did. The --ad-spdif-dtshd option will actually be deprecated in the next commit.