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* audio: define only a single NA speaker IDwm42015-05-073-13/+5
| | | | | 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
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* 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
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* ao_coreaudio: restore old physical format if format was changedwm42015-05-051-0/+16
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* 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
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* 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.
* 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.
* ao_null: add an option for testing channel layout selectionwm42015-04-271-2/+16
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* 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 Kurczewski2015-04-1315-75/+60
| | | | Signed-off-by: wm4 <wm4@nowhere>
* ao_coreaudio: fix inverted conditionwm42015-04-101-3/+4
| | | | And also use the correct type for the printf call below.
* 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
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* ao_wasapi: code formatting and alignmentKevin Mitchell2015-04-032-24/+23
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* audio: make all format query shortcuts macrosKevin Mitchell2015-04-034-7/+7
| | | | | 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
* ao_wasapi: abstract HRESULT_to_strKevin Mitchell2015-04-014-79/+67
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* ao_coreaudio: do not signal per-app volumewm42015-04-011-2/+0
| | | | | CoreAudio doesn't seem to have this concept. The volume is reset the next time audio is opened.
* ao_wasapi: remove redundant castsKevin Mitchell2015-03-313-42/+37
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* ao_wasapi: simplify hotplugKevin Mitchell2015-03-313-55/+22
| | | | | | | | | Take advantage of the fact that list_devs is called with a hotplug_inited ao. Also eliminate unnecessary nested function abstraction of hotplug_(un)init and list_devs. However, keep list_devs in ao_wasapi_utils.c since it uses the private functions get_device_id, get_device_name and exposing these would require including headers for IMMDevice in ao_wasapi_utils.h.
* ao_wasapi: fix device listingKevin Mitchell2015-03-313-92/+29
| | | | | remove depricated and convoluted validation. refer instead to the --audio-device option.
* ao/wasapi: add ao hotplugKevin Mitchell2015-03-315-64/+118
| | | | | | | Create a second copy of the change_notify structure for the hotplug ao. change_notify->is_hotplug distinguishes the hotplug version from the regular one monitoring the currently playing ao. Also make the change notification less verbose now that there might be two of them around.
* ao_alsa: add an option to ignore ALSA channel map negotiationwm42015-03-281-2/+6
| | | | This was requested, more or less.
* ao/wasapi: use built in KSDATAFORMATsKevin Mitchell2015-03-271-13/+8
| | | | | Rather than defining them ourselves. Thanks to rossy for figuring out the headers.
* ao/wasapi: add missing "if" bracesKevin Mitchell2015-03-262-35/+33
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* ao/wasapi: rewrite format searchKevin Mitchell2015-03-262-182/+300
| | | | | | | More clearly separate the exclusive and shared mode format discovery. Make the exclusive mode search more systematic in particular about channel maps (i.e., use chmap_sel). Assume that the same sample format / sample rates work for all channels to narrow the search space.
* ao_sndio: open device in blocking mode, don't inflate buffer artificiallyDmitrij D. Czarkoff2015-03-261-20/+2
| | | | | | The code actually uses blocking mode, so opening sound device in non-blocking mode results in choppy sound. Also, inflating the buffer isn't necessary in blocking mode, so the function may simply return without doing anything.
* ao_pulse: drop video role; fixes random mutingwm42015-03-241-1/+0
| | | | | | | | | The details are described in #1173. This "features" causes problems to users so often, it's better to remove it. Fixes #1173.
* ao: slightly extend debug messageswm42015-03-161-1/+4
| | | | | This function already got uglified with debug printing; might as well go all the way.
* ao: align audio buffer sizewm42015-03-131-0/+3
| | | | Might or might not matter.
* ao_coreaudio_exclusive: port to pull API, fix latency calculationswm42015-03-101-78/+37
| | | | | | | | | | | | | | Instead of maintaining a private ring buffer, use the generic support for audio APIs with pull callbacks (internally called AO pull API). This also fixes latency calculations: instead of just returning the ringbuffer status, the audio playback state is calculated better and includes interpolation. The main reason this wasn't done earlier was mid-stream format switching. The pull API can now handle it (in a way) by destroying and recreating the AO. This is a bit brutal, but quite simple. It's untested in this new AO, though. Some details might not be right, like how ot restores the old format when reloading.
* ao_coreaudio: move some helpers to utilswm42015-03-103-16/+20
| | | | Needed by ao_coreaudio_exclusive.c in the next commit.
* ao_coreaudio_exclusive: rip out pseudo volume controlwm42015-03-101-40/+1
| | | | | | | | | | | | | | | | | This could mute a digital passthrough stream by writing zeros. All other volume values did nothing. The comment about MPlayer dying hasn't been true in mpv for quite a while. It's even possible that it's fixed in upstream MPlayer. mpv will print a scary error message when trying to change volume with spdif, and continue normally. If we really want to mute by writing zeros, we should do it in a separate filter. But I'm not overly fascinated by this approach; is it even guaranteed receivers will not be confused by a stream of zeros? The main reason to remove this is that it's in the way of further cleanups.
* ao/wasapi: move resume to audio threadKevin Mitchell2015-02-233-24/+38
| | | | | | | | | | | | | | This echanges the two events hForceFeed/hFeedDone for hResume. This like the last commit makes things more deterministic. Importantly, the forcefeed is only done if there is not already a full buffer yet to be played by the device. This should fix some of the problems with exclusive mode. This commit also removes the necessity to have a proxy to the AudioClient object in the main thread. fixes #1529
* ao_wasapi: move reset into audio threadKevin Mitchell2015-02-232-9/+37
| | | | | | | | | This makes things a bit more deterministic. It ensures that the audio thread isn't doing anything between IAudioClient_Stop(), IAudioClient_Reset() and setting the sample_count to 0. Buffer overfilling on resume is still a problem in exclusive mode (see next commit).
* ao: fix null dereferenceStefano Pigozzi2015-02-141-0/+2
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* ao_coreaudio: add support for hotplug notificationsStefano Pigozzi2015-02-143-23/+96
| | | | | | | | | | This commit adds notifications for hot plugging of devices. It also extends the old behaviour of the `audio-out-detected-device` property which is now backed by the hotplugging code. This allows clients to be notified when the actual audio output device changes. Maybe hotplugging should be supported for ao_coreaudio_exclusive too, but it's device selection code is a bit fragile.
* ao_pulse: listen for hotplug eventswm42015-02-121-7/+41
| | | | | | | | | | | | | | | This requires jumping through multiple hoops on fire. Since the PulseAudio API is virtually undocumented, I'm not sure if this is correct either. We only react to sink events, and only to the NEW/REMOVE events. CHANGE events are ignored, because PulseAudio fires them far too often - even if the system is completely idle! If pa_sink_info.name can change, we're in trouble. pa_sink_info.description is not so important, but it'd also be a bit un-nice if it can change, and we don't update it. The weird way how the actual AO and the hotplug context share the same struct (ao) comes in handy here, although context_success_cb() still had to be duplicated from success_cb() - the unused argument has a different type.
* audio: add device change notification for hotpluggingwm42015-02-123-25/+120
| | | | | | | | | | | | | | | | | | | | | | | | | | | Not very important for the command line player; but GUI applications will want to know about this. This only adds the internal API; support for specific audio outputs comes later. This reuses the ao struct as context for the hotplug event listener, similar to how the "old" device listing API did. This is probably a bit unclean and confusing. One argument got reusing it is that otherwise rewriting parts of ao_pulse would be required (because the PulseAudio API requires so damn much boilerplate). Another is that --ao-defaults is applied to the hotplug dummy ao struct, which automatically applies such defaults even to the hotplug context. Notification works through the property observation mechanism in the client API. The notification chain is a bit complicated: the AO notifies the player, which in turn notifies the clients, which in turn will actually retrieve the device list. (It still has the advantage that it's slightly cleaner, since the AO stuff doesn't need to know about client API issues.) The weird handling of atomic flags in ao.c is because we still don't require real atomics from the compiler. Otherwise we'd just use atomic bitwise operations.
* ao: set correct client name when listing deviceswm42015-02-121-4/+3
| | | | | | | | | | This is a small oversight. The client name (as set on command line options or, more importantly, the client API) was not set when listing devices e.g. via the "audio-device-list" property. Might or might not fix #1578. Also adjust the log level for an unrelated message.
* ao_coreaudio: fix small memory leakStefano Pigozzi2015-02-031-0/+4
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* ao_coreaudio: use device UID instead of ID for selectionStefano Pigozzi2015-02-034-19/+46
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