summaryrefslogtreecommitdiffstats
path: root/audio
Commit message (Collapse)AuthorAgeFilesLines
* audio: simplify furtherwm42015-05-082-16/+8
| | | | | | Drop mp_chmap_diff() (which is unused too now), and implement mp_chmap_diffn() in a slightly simpler way. (Too bad there is no standard function for counting set bits.)
* audio: remove mp_chmap_contains()wm42015-05-082-12/+0
| | | | It's unsued now.
* ao: log reordered versions of channel mapswm42015-05-081-3/+10
| | | | Useful for debugging cases when no standard orders are used.
* audio: redo channel map fallback selectionwm42015-05-081-59/+27
| | | | | | | | | | | | | | | | | | | | Instead of somehow having 4 different cases with each their own weight, do it with a single function that decides which channel layout is the better fallback. This is simpler, and also introduces new (fixed) semantics. The new test added to test/chmap_sel.c actually works now. This is a mixed case with no perfect upmix or downmix, but the better choice is the one which loses the least channels from the original layout. One test also changes. If the input is 7.1(wide-side), and the available layouts are 7.1 and 5.1(side), the latter is now chosen instead of the former. This makes sense: both layouts contain 6 out of 8 channels from the original layout, but the 5.1(side) one is smaller. This follows the general logic. The 7.1 layout has FLC/RLC speakers instead of BL/BR, and judging by the names, "front left center" is completely different from "back left". If these should be exchangeable, a separate exception would have to be added.
* audio: add chmap utility functionwm42015-05-082-0/+10
|
* ao_alsa: log requested numbers of channels if ALSA rejects themwm42015-05-081-2/+3
|
* audio: fix messed up assert()wm42015-05-071-1/+1
| | | | This made no sense and always evaluated to true.
* audio: remove UNKNOWN pseudo speakerswm42015-05-072-11/+8
| | | | | | Reuse MP_SPEAKER_ID_NA for this. If all mp_chmap entries are set to NA, the channel layout has special "unknown channel layout" semantics, which are used to deal with some corner cases.
* audio: define only a single NA speaker IDwm42015-05-075-34/+16
| | | | | Remove the requirement from mp_chmap that speaker entries must be unique. Use this to get rid of all the redundant NA speaker IDs.
* ao_coreaudio_utils: don't list some formats as "unusable"wm42015-05-071-1/+1
| | | | | While mpv has no internal equivalent representation, they can still be used as physical CoreAudio formats. Thus this label is confusing.
* ao_sndio: add notice about padding channelswm42015-05-061-1/+3
| | | | (I won't do this, but someone else seeing this might.)
* ao_alsa: use new padding channels supportwm42015-05-061-21/+26
| | | | | | | | | | | | Sometimes, ALSA will return channel layouts with padded channels (NA speakers). Use them instead of failing. This still includes the old "braindeath" code to retry with a layout without NA channels. This might be helpful for performance, and also the padded channel layout string looks confusing. To be fair, I have not encountered a case yet which would really need this, and for which the old "braindeath" code did not fix it.
* ao_alsa: move ALSA -> mp channel map to a functionwm42015-05-061-11/+18
| | | | | One side effect is that the warning about too many channels goes away, and is replaced with printing the ALSA channel map as "unknown".
* ao_coreaudio_exclusive: check new format before waiting for changewm42015-05-061-12/+13
| | | | | It seems if the format was already set, setting the same format will not cause a property change.
* ao_coreaudio_exclusive: use atomics instead of volatilewm42015-05-061-19/+16
| | | | | | | | | | | volatile barely means anything. The polling is kind of bad too, but relatively harmless as device opening/closing is a rare event, and the format change is not expected to take long. Remove the pointless talloc call too (must have been a leftover from previous refactoring).
* ao_coreaudio_exclusive: rename "digital" -> "compressed"wm42015-05-061-22/+20
| | | | PCM is digital too.
* ao_coreaudio_exclusive: explicitly check for spdif formatswm42015-05-061-8/+5
|
* ao_coreaudio_exclusive: merge init_digital() functionwm42015-05-061-15/+3
| | | | | | No reason to keep them separate. It's an artifact from the old ao_coreaudio.c, which kept usage of two different APIs in the same file. Removes a forward reference too.
* ao_coreaudio_utils: decide formats by comparing raw bitswm42015-05-051-5/+6
| | | | | | | | | | | | | | | | Instead of trying to use af_format_conversion_score() (which tries to be all kinds of clever), just compare the raw bits as a quality measure. Do this because otherwise, weird formats like padded 24 bit formats will be excluded, even though they might be the highest precision formats for some hardware. This means that for now, the user would have to check whether the format is usable at all before calling ca_asbd_is_better(). But since this is currently only used for ao_coreaudio.c and for the physical format, it doesn't matter. If coreaudio-exclusive should get PCM support, the best would be to revert this change, and to add support for 24 bit formats directly.
* ao_coreaudio: log considered physical formatswm42015-05-051-0/+4
|
* ao_coreaudio: restore old physical format if format was changedwm42015-05-051-0/+16
|
* af: don't attempt to remove last filter for spdif filter removalwm42015-05-051-1/+1
| | | | | | | | | | | | | | | Some time ago, a mechanism was added for automatically removing PCM-only filters if the input format is spdif. This could cause an infinite loop if the AO did not support spdif, but was falling back to some PCM format. Then this code tried to remove the last filter, which is a dummy filter for receiving and queuing filter output. af_remove() simply fails gracefully in this case, so this happens over and over again. Fix by explicitly checking whether the filter to remove is a dummy filter. (af_remove() also fails only if the dummy filters are attempted to be removed - checking this directly is simpler.)
* audio: minor cosmeticswm42015-05-051-16/+16
| | | | | These ( ) were probably not removed when the format constants were changed from defines to an enum.
* ao_coreaudio: move channel mapping code to a separate filewm42015-05-053-255/+297
| | | | | | | | | | Move all of the channel map retrieval/negotiation code to a separate file. This will (probably) be helpful when extending ao_coreaudio_exclusive.c. Nothing else changes, other than some minor cosmetics and renaming, and changing some details for decoupling it from the ao_coreaudio.c internals.
* ao_coreaudio_utils: don't require talloc for fourcc_repr()wm42015-05-053-17/+13
| | | | | Instead, apply a trick to make the caller allocate enough space on the stack.
* ao_coreaudio_utils: unbreak default device selectionwm42015-05-051-4/+3
| | | | | | It appears this is the reason coreaudio-exclusive does not work without explicitly specifying a device, even if the default device maps to something passthrough-capable.
* ao_coreaudio_exclusive: fix latency calculation non-sensewm42015-05-051-1/+1
| | | | Didn't use the properties it was supposed to use.
* ao_coreaudio_utils: refine format selectionwm42015-05-051-19/+25
| | | | | | | | | | | Instead of always picking a somehow better format over the previous one, select a format that is equal to or better the requested format, but is also reasonably close. Drop the mFormatID comparison - checking the sample format handles this already. Make sure to exclude channel counts that can't be used.
* ao_coreaudio: change physical format before channel negotiationwm42015-05-051-4/+10
| | | | | | If for example the physical format is set to stereo, the reported multichannel layout will actually be stereo. It fixes itself only after the physical format is changed.
* ao_coreaudio: add an option for changing the physical formatwm42015-05-051-0/+55
| | | | | | | | | | | | ao_coreaudio uses AudioUnit - the OSX software mixer. In theory, it supports multichannel audio just fine. But in practice, this might be disabled by default, and the user is supposed to select a multichannel base format in the "Audio MIDI Setup" utility. This option attempts to change this setting automatically. Some possible disadvantages and caveats are listed in the manpage additions. It is off by default, since changing this might be rather bad behavior for a normal application.
* ao_coreaudio_utils: add a format negotiation helper functionwm42015-05-052-0/+37
|
* af_lavrresample: remove dead undefswm42015-05-051-3/+0
|
* ao_coreaudio: support padded channel layoutswm42015-05-051-2/+6
| | | | | | | If for example the audio settings are set to 5.1 output, but the hardware does 8 channels natively (HDMI), the reported channel layout will have 2 dummy channels. To avoid falling back to stereo, we have to write audio in this format to the device.
* audio: introduce support for padding channelswm42015-05-053-56/+142
| | | | | | | | | | | | | | | | | | | Some audio APIs explicitly require you to add dummy channels. These are not rendered, and only exist for the sake of the audio API or hardware strangeness. At least ALSA, Sndio, and CoreAudio seem to have them. This commit is preparation for using them with ao_coreaudio. The result is a bit messy. libavresample/libswresample don't have good API for this; avresample_set_channel_mapping() is pretty useless. Although in theory you can use it to add and remove channels, you can't set the channel counts. So we do the ordering ourselves by making sure the audio data is planar, and by swapping the plane pointers. This requires lots of messiness to get the conversions in place. Also, the input reordering is still done with the "old" method, and doesn't support padded channels - hopefully this will never be needed. (I tried to come up with cleaner solutions, but compared to my other attempts, the final commit is not that bad.)
* audio: introduce mp_audio readonly bitwm42015-05-042-1/+3
| | | | Convenience for the following commit.
* audio: chmap: explicitly drop channels not supported by lavcwm42015-05-041-2/+5
| | | | Basically as before, but avoid undefined behavior.
* audio: drop unused functionwm42015-05-042-10/+0
|
* ao_coreaudio: fix out of bounds accesswm42015-05-041-0/+2
| | | | | | ca_label_to_mp_speaker_id() checked whether the last entry was >= 0, but actually this condition was never true, and MP_SPEAKER_ID_UNKNOWN0 is not negative.
* ao_coreaudio_exclusive: check format explicitly on change notifcationwm42015-04-291-6/+11
| | | | | | | | | This should for now be equivalent; it's merely more explicit and will be required if we add PCM support. Note that the property listeners actually tell you what property exactly changed, but resolving the current listener mess would be too hard. So check for changes manually.
* ao_coreaudio_utils: log mp format with CoreAudio format descriptionwm42015-04-291-2/+4
| | | | As a consequence, it also logs whether mpv can a this format at all.
* ao_coreaudio_utils: add function for ASBD -> mp format lookupwm42015-04-292-7/+59
| | | | | | | | | | Useful with some of the following commits. ca_fill_asbd() should behave exactly as before. Instead of actually implementing the inverse function of ca_fill_asbd(), just loop over the (small) list of mpv functions and check if any mpv equivalent to a given ASBD exists.
* ao_coreaudio_utils: float is not a signed integer formatwm42015-04-291-3/+3
| | | | | | | | | kAudioFormatFlagIsSignedInteger implicates that it's only used with integer formats. The mpv internal flag on the other hand signals the presence of a sign, and this is set on float formats. Until now, this probably worked fine, because at least AudioUnit is ignoring the uncorrect flag.
* ao_coreaudio_exclusive: move code for getting original formatwm42015-04-281-6/+4
| | | | | Should be almost equivalent, unless there are streams on which this call does not work for unknown reasons.
* ao_coreaudio_utils: change audio format loggingwm42015-04-281-3/+3
| | | | Make it easier to distinguish the fields.
* ao_coreaudio_exclusive: account for additional latencywm42015-04-281-3/+10
| | | | | | | Whether this is correct is unknown. This change tripples the latency from ~15ms to ~45ms. XBMC does this, VLC does not from what I could see.
* audio: separate fallbacks for upmix and downmix caseswm42015-04-281-12/+18
| | | | | | | | We always want to prefer upmix to downmix, as long as it makes sense. Even if the upmix is not "perfect" (not just adding channels), we want to prefer the upmix. Cleanup for commit d3c7fd9d.
* audio: avoid downmixing in a certain special-casewm42015-04-271-3/+3
| | | | | | | | As indicated by the added test. In this case, fallback and downmix have the same score, but fallback happens to give better results. So prefer fallback over downmix. (This is probably not a correct solution.)
* ao_null: add an option for testing channel layout selectionwm42015-04-271-2/+16
|
* player: change video-bitrate and audio-bitrate propertieswm42015-04-201-3/+0
| | | | | | | | | | | | | | Remove the old implementation for these properties. It was never very good, often returned very innaccurate values or just 0, and was static even if the source was variable bitrate. Replace it with the implementation of "packet-video-bitrate". Mark the "packet-..." properties as deprecated. (The effective difference is different formatting, and returning the raw value in bits instead of kilobits.) Also extend the documentation a little. It appears at least some decoders (sipr?) need the AVCodecContext.bit_rate field set, so this one is still passed through.
* af_lavrresample: fix drainingwm42015-04-181-8/+8
| | | | | configure_lavrr() clears s->pending, so we have to assign it after that call.
* ao_alsa: fallback to stereo channel layout if everything else failswm42015-04-141-1/+4
| | | | | | mp_chmap_from_channels_alsa() doesn't always succeed - there are a bunch of channel counts for which no defined ALSA layout exists. Fallback to stereo in this case. (Normally, this code path shouldn't happen at all.)
* Update license headersMarcin Kurczewski2015-04-1357-288/+232
| | | | Signed-off-by: wm4 <wm4@nowhere>
* af_lavrresample: minor simplificationwm42015-04-121-4/+4
| | | | | The in/out pointers usually have not much meaning outside of AF_CONTROL_REINIT. Also remove the redundant casts.
* af_lavrresample: allow resetting output sample formatwm42015-04-121-2/+3
| | | | It must be allowed to set format==0.
* audio/filter: fully renegotiate audio formats on every reconfigwm42015-04-121-0/+10
| | | | | | | | | | | | | | | | | | | It could happen that a lavrresample filter would keep its old output format when the decoder changed its output format. This simply happened because the output format was never reset. Normally, this was not an issue, because lavrresample filters only inserted for format conversion were removed on format changes. But if --no-audio-pitch-correction is set and playback speed is changed, then there is a "permanent" lavrresample filter in the filter chain, which shows this behavior. Fix by explicitly resetting output formats for all filters which support it. Note: this can crash with libswresample in some cases. I'm not sure if this is mpv's fault or libswresample's, but since it works with libavresample, I'm going to assume it's not our's.
* ao_coreaudio: fix inverted conditionwm42015-04-101-3/+4
| | | | And also use the correct type for the printf call below.
* audio: automatically deatch filters if spdif prevents their usewm42015-04-072-5/+17
| | | | Fixes #1743 and partially #1780.
* audio: change a detail about filter insertionwm42015-04-073-10/+14
| | | | | | | | | | The af_add() function has a problem: if the inserted filter returns AF_DETACH during init, the function will have a dangling pointer. Until now this was avoided by making sure none of the used filters actually return AF_DETACH, but it's getting infeasible. Solve this by requiring passing an unique label to af_add(), which is then used instead of the pointer.
* ao_alsa: change log outputwm42015-04-071-12/+15
| | | | | | | | Silence the usually user-visible warning about unsupported channel maps. This might be an ALSA bug, but ALSA will never fix this behavior anyway. (Or maybe it's a feature.) Log some other information that might be useful.
* ao_coreaudio: do not error if retrieving info for verbose mode failswm42015-04-071-6/+6
| | | | | | The message log level shouldn't get to decide whether something fails or not. So replace the fatal error check on the verbose output code path with a warning.
* ao/wasapi: use atomic state variable instead of different eventsKevin Mitchell2015-04-044-65/+78
| | | | | | | | | Unfortunately, because we have proxy objects (pAudioVolumeProxy, pEndpointVolumeProxy, pSessionControlProxy) it looks like we still have to use MsgWaitForMultipleObjects and watch for and dispatch pending messages: https://msdn.microsoft.com/en-us/library/windows/desktop/ms680112%28v=vs.85%29.aspx
* ao/wasapi: reorder priv membersKevin Mitchell2015-04-041-12/+14
|
* ao_wasapi: code formatting and alignmentKevin Mitchell2015-04-032-24/+23
|
* audio: make all format query shortcuts macrosKevin Mitchell2015-04-039-25/+15
| | | | | af_fmt_is_float and af_fmt_is_planar were previously inconsistent with AF_FORAMT_IS_SPECIAL/AF_FORMAT_IS_IEC61937
* ao_wasapi: passthrough reworkKevin Mitchell2015-04-032-161/+152
| | | | | | | | | | | | | | | * unify passthrough and pcm exclusive mode format setting/testing * set passthrough format parameters correctly * support all of mpv's existing passthrough formats * automatically test passthrough with exclusive mode and enable exclusive if it succeeds, even if it was not explictly requested. this obviates the need for --ao=wasapi,wasapi=exclusive * if passthrough fails (such as the device doesn't support the format), fallback to either exclusive pcm or shared mode depending on what the user specified. Right now this isn't very useful as it still fails due to the decoder path remainin stuck on spdif. fixes #1742
* af_lavrresample: always normalize (libswresample is stupid)wm42015-04-021-0/+4
| | | | | | | | | | libswresample doesn't normalize when remixing to a float format. This will cause clipping due to float samples being out of the allowed range. Fortunately this extremely bad default can be changed. This does not happen with libavresample: it normalizes by default. Fixes #1752.
* af: remove unused functionswm42015-04-012-34/+0
|
* ao_wasapi: abstract HRESULT_to_strKevin Mitchell2015-04-014-79/+67
|
* mixer: per-app volume and private volume conflictwm42015-04-011-1/+3
| | | | | | Per-app volume would change the volume across all instances of the same application, while a private volume control (HAS_PER_APP_VOLUME) obviously should influence only one instance/audio stream only.
*