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* ao_alsa: fix channel map in pre-channel map API casewm42014-11-251-0/+1
| | | | Forgotten in commit 5d5f5b09.
* ao_alsa: always enable "plug" plugin for non-default devicewm42014-11-251-3/+2
| | | | | | | | | | | | | | This seems safer: otherwise, opening the AO could randomly fail if the audio formats happens to be not float. Unfortunately, this only works if the user does not select a device. Since ALSA devices are arbitrary strings, including plugins with complex parameters, it's not trivial or maybe even impossible to edit the string in a way the "plug" plugin is added. With --audio-device, it would be safe for users to select either "default" or one of the "plughw" devices. Everything else seems questionable.
* ao_alsa: select and set channel maps via channel map APIwm42014-11-251-28/+125
| | | | | | | | | | | | | | | | Use the ALSA channel map API for querying and selecting supported channel maps. Since we (probably?) want to be compatible with ALSA versions before the change, we still try to select the device name by channel map, and open that device. There's no way to negotiate a channel map before opening, so we're stuck with this approach. Fortunately, it seems these devices allow selecting and setting any other supported channel layout, so maybe this is not an issue at all. In particular, this avoids selecting the default (dmix) device, which can only do stereo. Most code is based on Martin Herkt <lachs0r@srsfckn.biz>'s alsa_ng branch, with heavy modifications.
* ao_alsa: minor fixeswm42014-11-251-4/+6
| | | | | | | | | | | | | Don't crash if no fallback channel layout could be found (caller can't handle NULL return from select_chmap()). Apparently this could never actually happen, though. Don't treat snd_pcm_hw_params_set_periods_near() failure as fatal error. Same deal as with snd_pcm_hw_params_set_buffer_time_near(). Actually free channel maps returned by snd_pcm_get_chmap(). Adjust some messages.
* ao_alsa: cleanupswm42014-11-251-97/+57
| | | | | | | | | No functional changes. ALSA_PCM_NEW_HW_PARAMS_API was a pre-ALSA 1.0.0 thing and does nothing with modern ALSA. It stopped being necessary about 10 years ago. 3 functions are moved to avoid forward references.
* audio: make mp_chmap_to_str() return a stack-allocated stringwm42014-11-241-6/+2
| | | | Simplifies memory management.
* ao_alsa: try to use the channel map reported by ALSAwm42014-11-241-0/+64
| | | | | | | | If ALSA reports a channel map, and it looks like it makes sense (i.e. could be converted to mpv channel map, and the channel count matches), then use that instead of the channel map we are assuming. This is based on code written by lachs0r (alsa_ng branch).
* ao_alsa: check for EAGAIN toowm42014-11-171-1/+1
| | | | | | | Simply retry on EAGAIN. I've seen this in several other projects; it might be just cargo-culting though.
* audio/out: consistently use double return type for get_delaywm42014-11-091-5/+5
| | | | | ao_get_delay() returns double, but the get_delay callback still returned float.
* ao_alsa: don't make snd_pcm_hw_params_set_buffer_time_near() error fatalwm42014-10-311-1/+7
| | | | | | | | | | | | | Apparently this can "sometimes" return an error. In my opinion, this should never return an error: neither the semantics of the function, nor the ALSA documentation or ALSA sample code seem to indicate that a failure is to be expected. I'm not perfectly sure about this though (I blame ALSA being a weird, big, underdocumented API). Since it causes problems for some users, and since there is really no reason why we should abort on such an error, turn it into a warning. Fixes #1231.
* ao_alsa: move parameter append code to a functionwm42014-10-231-16/+27
| | | | | Why not. (I thought I needed this, but my other experiments failed. So this is merely a minor cleanup.)
* audio: change internal device listing APIwm42014-10-101-2/+2
| | | | | Now we run ao_driver->list_devs on a dummy AO instance, which will probably confuse everyone. This is done for the sake of PulseAudio.
* ao_alsa: implement device listing & selectionwm42014-10-091-0/+27
| | | | | | | Unfortunately, ALSA is particularly bad with this, because mpv has to add all sorts of magic crap to the device name to make things work. The device selection overrides this, so explicitly selecting devices will most likely break your audio. This has yet to be solved.
* audio: cleanup spdif format definitionswm42014-09-231-11/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Before this commit, there was AF_FORMAT_AC3 (the original spdif format, used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS and DTS-HD), which was handled as some sort of superset for AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used IEC61937-framing, but still was handled as something "separate". Technically, all of them are pretty similar, but may use different bitrates. Since digital passthrough pretends to be PCM (just with special headers that wrap digital packets), this is easily detectable by the higher samplerate or higher number of channels, so I don't know why you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs. AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is just a mess. Simplify this by handling all these formats the same way. AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3). All AOs just accept all spdif formats now - whether that works or not is not really clear (seems inconsistent due to earlier attempts to make DTS-HD work). But on the other hand, enabling spdif requires manual user interaction, so it doesn't matter much if initialization fails in slightly less graceful ways if it can't work at all. At a later point, we will support passthrough with ao_pulse. It seems the PulseAudio API wants to know the codec type (or maybe not - feeding it DTS while telling it it's AC3 works), add separate formats for each codecs. While this reminds of the earlier chaos, it's stricter, and most code just uses AF_FORMAT_IS_IEC61937(). Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to include special formats, so that it always describes the fundamental sample format type. This also ensures valid AF formats are never 0 (this was probably broken in one of the earlier commits from today).
* audio: drop swapped-endian audio formatswm42014-09-231-18/+14
| | | | | | | | | | | | | | | | | | | | Until now, the audio chain could handle both little endian and big endian formats. This actually doesn't make much sense, since the audio API and the HW will most likely prefer native formats. Or at the very least, it should be trivial for audio drivers to do the byte swapping themselves. From now on, the audio chain contains native-endian formats only. All AOs and some filters are adjusted. af_convertsignendian.c is now wrongly named, but the filter name is adjusted. In some cases, the audio infrastructure was reused on the demuxer side, but that is relatively easy to rectify. This is a quite intrusive and radical change. It's possible that it will break some things (especially if they're obscure or not Linux), so watch out for regressions. It's probably still better to do it the bulldozer way, since slow transition and researching foreign platforms would take a lot of time and effort.
* audio: remove swapped-endian spdif formatswm42014-09-231-6/+4
| | | | | | | | | | | | | | | | | | | | | | IEC 61937 frames should always be little endian (little endian 16 bit words). I don't see any apparent need why the audio chain should handle swapped-endian formats. It could be that some audio outputs might want them (especially on big endian architectures). On the other hand, it's not clear how that works on these architectures, and it's not even known whether the current code works on big endian at all. If something should break, and it should turn out that swapped-endian spdif is needed on any platform/AO, swapping still could be done in-place within the affected AO, and there's no need for the additional complexity in the rest of the player. Note that af_lavcac3enc outputs big endian spdif frames for unknown reasons. Normally, the resulting data is just pulled through an auto- inserted conversion filter and turned into little endian. Maybe this was done as a trick so that the code didn't have to byte-swap the actual audio frame. In any case, just make it output little endian frames. All of this is untested, because I have no receiver hardware.
* audio/out: always round get_space on period sizewm42014-09-061-1/+1
| | | | | | | | | | | Round get_space() results in the same way play() rounds the input size. Some audio APIs do this for various reasons. This affects only "push" based AOs. Some of these need no change, because they either do it already right (like ao_openal), or they seem not to have any such requirements (like ao_pulse). Needed for the following commit.
* ao_alsa: disable use of non-interleaved formats by defaultwm42014-07-301-0/+6
| | | | | | | | Some ALSA plugins take non-interleaved audio, but treat it as interleaved, which results in various funny bugs. Users keep hitting this issue, and it just doesn't seem worth the trouble. CC: @mpv-player/stable
* Audit and replace all ctype.h useswm42014-07-011-1/+0
| | | | | | | | | | | | | | | | Something like "char *s = ...; isdigit(s[0]);" triggers undefined behavior, because char can be signed, and thus s[0] can be a negative value. The is*() functions require unsigned char _or_ EOF. EOF is a special value outside of unsigned char range, thus the argument to the is*() functions can't be a char. This undefined behavior can actually trigger crashes if the implementation of these functions e.g. uses lookup tables, which are then indexed with out-of-range values. Replace all <ctype.h> uses with our own custom mp_is*() functions added with misc/ctype.h. As a bonus, these functions are locale-independent. (Although currently, we _require_ C locale for other reasons.)
* Add more constwm42014-06-111-1/+1
| | | | | | | While I'm not very fond of "const", it's important for declarations (it decides whether a symbol is emitted in a read-only or read/write section). Fix all these cases, so we have writeable global data only when we really need.
* ao_alsa: make device the first sub optionwm42014-05-311-1/+1
| | | | This is more convenient.
* ao_alsa: reduce spurious wakeupswm42014-05-301-8/+14
| | | | | | Apparently this can happen. So actually only return from waiting if ALSA excplicitly signals that new output is available, or if we are woken up externally.
* ao_alsa: use poll() to wait for devicewm42014-05-301-0/+30
| | | | | This means the audio feed thread is woken up exactly at the time new data is needed, instead of using a time-based heuristic.
* af_lavrresample: remove avresample_set_channel_mapping() fallbackswm42014-03-161-1/+0
| | | | | | | This function is now always available. Also remove includes of reorder_ch.h from some AOs (these are just old relicts).
* ao_alsa: reduce default buffer sizewm42014-03-101-1/+1
| | | | | | | | | | In general, we don't need to have a large hw audio buffer size anymore, because we can quickly fill it from the soft buffer. Note that this probably doesn't change much anyway. On my system (dmix enabled), the buffer size is only 170ms, and ALSA won't give more. Even when using a hardware device the buffer size seems to be limited to 341ms.
* ao_alsa: fix return value for volume operations with spdifwm42014-03-101-1/+1
| | | | | | | | | | | This AO pretended to support volume operations when in spdif passthrough mode, but actually did nothing. This is wrong: at least the GET operations must write their argument. Signal that volume is unsupported instead. This was probably a hack to prevent insertion of volume filters or so, but it didn't work anyway, while recovering after failed volume filter insertion does work, so this is not needed at all.
* ao_alsa: remove unneeded initializationswm42014-03-091-6/+0
| | | | priv is 0-initialized, can_pause is always overwritten later.
* ao_alsa: check ALSA PCM state before pause and resumefoo862014-03-091-5/+9
| | | | | | | | | | It is possible to have ao->reset() called between ao->pause() and ao->resume() when seeking during the pause. If the underlying PCM supports pausing, resuming an already reset PCM will produce an error. Avoid that by explicitly checking PCM state before calling snd_pcm_pause(). Signed-off-by: wm4 <wm4@nowhere>
* audio/out: make draining a separate operationwm42014-03-091-6/+10
| | | | | | | | | | | | Until now, this was always conflated with uninit. This was ugly, and also many AOs emulated this manually (or just ignored it). Make draining an explicit operation, so AOs which support it can provide it, and for all others generic code will emulate it. For ao_wasapi, we keep it simple and basically disable the internal draining implementation (maybe it should be restored later). Tested on Linux only.
* audio/out: make ao struct opaquewm42014-03-091-0/+1
| | | | | | We want to move the AO to its own thread. There's no technical reason for making the ao struct opaque to do this. But it helps us sleep at night, because we can control access to shared state better.
* ao_alsa: remove 9 year old typowm42014-01-021-1/+0
| | | | | Actually, remove the whole comment, because it's outdated and get_space() returns the number of free samples now.
* ao_alsa: Unbreak pause/resumeMartin Herkt2014-01-021-11/+10
| | | | Well that was dumb.
* ao_alsa: Fix PCM resume after suspendMartin Herkt2014-01-021-15/+15
| | | | Fixes #324
* ao_alsa: don't set ALSA message callbackwm42013-12-211-21/+0
| | | | | | | This could output additional, potentially useful error messages. But the callback is global and not library-safe, and would require us to add additional state. Remove it, because it's obviously too much of a pain. Also, it seems ALSA prints stuff to stderr anyway.
* Split mpvcore/ into common/, misc/, bstr/wm42013-12-171-1/+1
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* Move options/config related files from mpvcore/ to options/wm42013-12-171-2/+2
| | | | | | | | | Since m_option.h and options.h are extremely often included, a lot of files have to be changed. Moving path.c/h to options/ is a bit questionable, but since this is mainly about access to config files (which are also handled in options/), it's probably ok.
* ao_alsa: remove unneeded checkswm42013-11-301-19/+11
| | | | | If initialization succeeds, p->alsa should always be set. Additional checks are not needed, and also this wasn't even done consistently.
* ao_alsa: enable "plug" for non-interleaved float formats toowm42013-11-301-2/+1
| | | | | | I have no idea what this code does, but it seems logical it should be active for all float formats, not just for float with interleaved access.
* ao_alsa: disable ALSA resampling by default againwm42013-11-291-0/+9
| | | | | | | | | | | | | | | | | | | | | This partially reverts commit 7d152965. It turns out that at least some ALSA drivers (at least snd-hda-intel) report incorrect audio delay with non-native sample rates, even if the sample rate is only very slightly different from the native one. For example, 48000Hz is fine on my hda-intel system, while both 8000Hz and 47999Hz lead to a delay off by 40ms (according to mpv's A/V difference display), which suggests that something in ALSA is calculating the delay using the wrong sample rate. As an additional problem, with ALSA resampling enabled, using 48001Hz/float/2ch fails, while 49000Hz/float/2ch or 48001Hz/s16/2ch work. With resampling disabled, all these cases work obviously, because our own resampler doesn't just refuse any of these formats. Since some people want to use the ALSA resampler (because it's highly configurable, supports multiple backends, etc.), we still allow enabling ALSA resampling with an ao_alsa suboption.
* ao_alsa: do not forcibly disable ALSA resamplingMartin Herkt2013-11-261-6/+0
| | | | | | | | Resampling with non-ancient ALSA setups works fine, so there is no need to keep this around. Furthermore, as of writing, the default builtin resampler used by many ALSA setups (taken from libspeex) actually has higher quality than the default resampling modes of avresample and swresample.
* audio: drop "_NE"/"ne" suffix from audio formatswm42013-11-151-2/+2
| | | | | | You get the native format by not appending any suffix to the format. This change includes user-facing names, e.g. for the --format option.
* ao_alsa: non-interleaved access is not always availablewm42013-11-141-0/+5
| | | | | | I thought this would always work... how disappointing. Revert to interleaved format if requesting non-interleaved fails.
* ao_alsa: support non-interleaved audiowm42013-11-121-25/+23
| | | | | | | | ALSA supports non-interleaved audio natively using a separate API function for writing audio. (Though you have to tell it about this on initialization.) ALSA doesn't have separate sample formats for this, so just pretend to negotiate the interleaved format, and assume that all non-interleaved formats have an interleaved companion format.
* audio/out: prepare for non-interleaved audiowm42013-11-121-11/+5
| | | | | | | | | | | | | | | | | | | This comes with two internal AO API changes: 1. ao_driver.play now can take non-interleaved audio. For this purpose, the data pointer is changed to void **data, where data[0] corresponds to the pointer in the old API. Also, the len argument as well as the return value are now in samples, not bytes. "Sample" in this context means the unit of the smallest possible audio frame, i.e. sample_size * channels. 2. ao_driver.get_space now returns samples instead of bytes. (Similar to the play function.) Change all AOs to use the new API. The AO API as exposed to the rest of the player still uses the old API. It's emulated in ao.c. This is purely to split the commits changing all AOs and the commits adding actual support for outputting N-I audio.
* audio/out: reject non-interleaved formatswm42013-11-121-0/+2
| | | | | | | | | | No AO can handle these, so it would be a problem if they get added later, and non-interleaved formats get accepted erroneously. Let them gracefully fall back to other formats. Most AOs actually would fall back, but to an unrelated formats. This is covered by this commit too, and if possible they should pick the interleaved variant if a non-interleaved format is requested.
* ao: add ao_play_silence, use for ao_alsa and ao_osswm42013-11-101-6/+2
| | | | | Also add a corresponding function to audio/format.c, which fills an audio block with silence.
* ao_alsa: use correct magic spdif flagswm42013-11-091-14/+11
| | | | | I accidentally copied the AES4/ORIGFS constants from the ALSA headers, instead of the AES3/FS ones. The difference is probably important.
* ao: print requested audio format on initwm42013-11-091-3/+0
| | | | Also remove the rather bad/incomplete log calls from ao_alsa and ao_oss.
* ao_alsa: add magic spdif parameterswm42013-11-091-1/+25
| | | | | | | | | | | | | | I have no idea what these do, but apparently they are needed to inform ALSA about spdif configuration. First, replace the literal constant "6" for the AES0 parameter with the symbolic constants from the ALSA headers (the final value is the same). Second, copy paste some funky looking parameter setup from VLC's alsa output for setting the AES1, AES2, AES3 parameters. (The code is actually not literally copy-pasted, but does exactly the same.) My small but non-zero hope is that this could make DTS-HD work, or at least work into that direction. I can't test spdif stuff though, and for DTS-HD not even opening the ALSA device succeeds on my system.
* ao_alsa: redo the way parameters are added in the spdif casewm42013-11-091-38/+24
| | | | | | | | | | | Using spdif with alsa requires adding magic parameters to the device name, and the existing code tried to deal with the situation when the user wanted to add parameters too. Rewrite this code, in particular remove the duplicated parameter string as preparation for the next commit. The new code is a bit stricter, e.g. it doesn't skip spaces before and after '{' and '}'. (Just don't add spaces.)
* audio: replace af_fmt2str_short -> af_fmt_to_strwm42013-11-071-1/+1
| | | | Also, remove all af_fmt2str usages.
* ao_alsa: return negative value on error in play()wm42013-10-301-3/+3
| | | | | No functional change, because the only user of ao_play() ignores return values below 1.
* ao_alsa: don't include alloca.hwm42013-10-251-1/+0
| | | | | | | | It's true that ALSA uses alloca() in some of its API functions, but since this is hidden behind macros in the ALSA headers, we have no reason to include alloca.h ourselves. Might help with portability (FreeBSD).
* audio/out: remove useless info struct and redundant fieldswm42013-10-231-6/+3
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* mixer: make struct opaquewm42013-09-201-1/+0
| | | | Also remove stray include statements from ao_alsa and ao_oss.
* audio/out: do some mp_msg conversionswm42013-08-221-54/+35
| | | | | | | 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.
* core: move contents to mpvcore (2/2)