summaryrefslogtreecommitdiffstats
path: root/audio
Commit message (Collapse)AuthorAgeFilesLines
* threads: use utility+POSIX functions instead of weird wrapperswm42015-05-151-3/+6
| | | | | | | | | There is not much of a reason to have these wrappers around. Use POSIX standard functions directly, and use a separate utility function to take care of the timespec calculations. (Course POSIX for using this weird format for time values.) (cherry picked from commit 92b9d75d7256be71d8c8b18438af9494b78f0e96)
* ao: make better use of atomicswm42015-05-152-14/+11
| | | | | | | The main reason for this was compatibility; but some associated problems have been solved in the previous commit. (cherry picked from commit ca9964a4fb6b1faa0155da43b3c815db0075e2d5)
* audio: simplify furtherwm42015-05-092-16/+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.) (cherry picked from commit 00130651dac758f90bf98306a9d1e569ed4155ca)
* audio: remove mp_chmap_contains()wm42015-05-092-12/+0
| | | | | | It's unsued now. (cherry picked from commit 8d5924f2c9c7d80b45cd68b44cb9c74e7b0b5a8c)
* ao: log reordered versions of channel mapswm42015-05-091-3/+10
| | | | | | Useful for debugging cases when no standard orders are used. (cherry picked from commit 8b7035c8ff15f14c17b6c019e951226b9eeaca02)
* audio: redo channel map fallback selectionwm42015-05-091-59/+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. (cherry picked from commit 3560a50029e160f0606d0cdc6aa1da662bbcace8)
* audio: remove UNKNOWN pseudo speakerswm42015-05-092-11/+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. (cherry picked from commit 55e777f10b3e241f2634b471e482bab230773ce0)
* audio: define only a single NA speaker IDwm42015-05-095-36/+17
| | | | | | | Remove the requirement from mp_chmap that speaker entries must be unique. Use this to get rid of all the redundant NA speaker IDs. (cherry picked from commit b91b4944bd7ddf6fef4c4254d457117017292c0a)
* audio: add chmap utility functionwm42015-05-092-0/+10
| | | | (cherry picked from commit d32b71d52e9a45c141d2bd132189db68613ab0fb)
* ao_alsa: log requested numbers of channels if ALSA rejects themwm42015-05-091-2/+3
| | | | (cherry picked from commit ad9bce2a5ca62f6a64f65fe79ae170edc0e05da4)
* audio: fix messed up assert()wm42015-05-081-1/+1
| | | | | | This made no sense and always evaluated to true. (cherry picked from commit 7b09654c33ca81aede475235121ebc938791dc80)
* ao_coreaudio_utils: don't list some formats as "unusable"wm42015-05-081-1/+1
| | | | | | | While mpv has no internal equivalent representation, they can still be used as physical CoreAudio formats. Thus this label is confusing. (cherry picked from commit 1bcb82ec93cc3e037df2dd4e2216a473fe87baf9)
* ao_sndio: add notice about padding channelswm42015-05-071-1/+3
| | | | | | (I won't do this, but someone else seeing this might.) (cherry picked from commit cd5ab98ff992217abfd0234601c21eb0fe0dbc19)
* ao_alsa: use new padding channels supportwm42015-05-071-21/+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. (cherry picked from commit 85fc6b2a0569b24c5652f600d90d7a131b61eb07)
* ao_alsa: move ALSA -> mp channel map to a functionwm42015-05-071-11/+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". (cherry picked from commit d577872a28c9729e987566530905bde238af8109)
* ao_coreaudio_exclusive: check new format before waiting for changewm42015-05-071-12/+13
| | | | | | | It seems if the format was already set, setting the same format will not cause a property change. (cherry picked from commit 0ae0e90eb5348c58d5b4f13fe0792199c460a4b6)
* ao_coreaudio_exclusive: use atomics instead of volatilewm42015-05-071-19/+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). (cherry picked from commit 4444ff48fa578461688fe9feb9ebcd996cd64506)
* ao_coreaudio_exclusive: rename "digital" -> "compressed"wm42015-05-071-22/+20
| | | | | | PCM is digital too. (cherry picked from commit 028739932bf4e2d32439b3756811a2b06cc81128)
* ao_coreaudio_exclusive: explicitly check for spdif formatswm42015-05-071-8/+5
| | | | (cherry picked from commit 1e1045b13ea4acbbd77dd52c4e0599f1517e6ac3)
* ao_coreaudio_exclusive: merge init_digital() functionwm42015-05-071-15/+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. (cherry picked from commit 32bc61ae07fe441c327b4aa96dd80fa4771fd569)
* ao_coreaudio_utils: decide formats by comparing raw bitswm42015-05-071-5/+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. (cherry picked from commit 4ffcf2531bb525c19c3b6df75ecb27c5cffbdd28)
* af: don't attempt to remove last filter for spdif filter removalwm42015-05-071-1/+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.) (cherry picked from commit 0025030cef757327769982333f9105aa510c393d)
* audio: minor cosmeticswm42015-05-071-16/+16
| | | | | | | These ( ) were probably not removed when the format constants were changed from defines to an enum. (cherry picked from commit d76f9a484ea7795655637eb0ddc8655aa4fff345)
* ao_coreaudio_utils: don't require talloc for fourcc_repr()wm42015-05-073-17/+13
| | | | | | | Instead, apply a trick to make the caller allocate enough space on the stack. (cherry picked from commit 399267393bb96710cde53c2fc7563f55cc32deb8)
* ao_coreaudio_utils: unbreak default device selectionwm42015-05-071-4/+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. (cherry picked from commit 7a5f5a8adf5921ed8fcee29d76113d9a7f018974)
* ao_coreaudio_exclusive: fix latency calculation non-sensewm42015-05-071-1/+1
| | | | | | Didn't use the properties it was supposed to use. (cherry picked from commit bbedceb467033b239b35ee9b2db963a93d8a57c9)
* ao_coreaudio_utils: refine format selectionwm42015-05-071-19/+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. (cherry picked from commit fd6809f98a546c2abe87b378bb1fe0bbec40a4ef)
* ao_coreaudio_utils: add a format negotiation helper functionwm42015-05-072-0/+37
| | | | (cherry picked from commit 305a85cc9aa169a75317acb55e539f49d420f629)
* ao_coreaudio: support padded channel layoutswm42015-05-071-2/+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. (cherry picked from commit 4d8a7e03944155bf07ba9a775cf9554bb1c76f0f)
* audio: introduce support for padding channelswm42015-05-073-56/+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.) (cherry picked from commit 06050aed9906b784159ad03e86e13348c4d9fa47)
* audio: introduce mp_audio readonly bitwm42015-05-072-1/+3
| | | | | | Convenience for the following commit. (cherry picked from commit 1b0b094ca2c25ad162f8f8c84ebebef9a963552e)
* audio: chmap: explicitly drop channels not supported by lavcwm42015-05-071-2/+5
| | | | | | Basically as before, but avoid undefined behavior. (cherry picked from commit 937c8e513f7b948fff0746e80ecf3d27d7007abe)
* audio: drop unused functionwm42015-05-072-10/+0
| | | | (cherry picked from commit 548cd826c24b7f56b597785f0b83a47cbf4a0465)
* ao_coreaudio: fix out of bounds accesswm42015-05-071-0/+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. (cherry picked from commit eead97f10303436b8da1c75dcdaa79efaba5b015)
* ao_coreaudio_exclusive: check format explicitly on change notifcationwm42015-05-071-6/+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. (cherry picked from commit 382434d45a72967f5b607c871e363e02dce1f1e6)
* ao_coreaudio_utils: log mp format with CoreAudio format descriptionwm42015-05-071-2/+4
| | | | | | As a consequence, it also logs whether mpv can a this format at all. (cherry picked from commit 34a5229b231f15c95876fed472bd1edc5283db31)
* ao_coreaudio_utils: add function for ASBD -> mp format lookupwm42015-05-072-7/+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. (cherry picked from commit 32b835c03b4dc98a0344d171adef36c7562f1e7b)
* ao_coreaudio_utils: float is not a signed integer formatwm42015-05-071-3/+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. (cherry picked from commit 3295ce48ab4badff0e13e2e9c2a1ec945413d4e2)
* ao_coreaudio_exclusive: move code for getting original formatwm42015-04-291-6/+4
| | | | | | | Should be almost equivalent, unless there are streams on which this call does not work for unknown reasons. (cherry picked from commit 8b4ca5806207c1482df30d9815e6970697cea5b2)
* ao_coreaudio_utils: change audio format loggingwm42015-04-291-3/+3
| | | | | | Make it easier to distinguish the fields. (cherry picked from commit d5e9bf66a1e0c4578bd8bef5c9f725dbc47e9fc6)
* ao_coreaudio_exclusive: account for additional latencywm42015-04-291-3/+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. (cherry picked from commit 5f86fad2f0ab76b7497230b18cd146a7c4d38cd2)
* audio: separate fallbacks for upmix and downmix caseswm42015-04-291-12/+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. (cherry picked from commit c4aa13615501189c55c23448d436074e5f92c8cc)
* audio: avoid downmixing in a certain special-casewm42015-04-281-3/+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.) (cherry picked from commit d3c7fd9d7c971086a3d6fde5f6f1bc4ef0b2e904)
* 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