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* ao: use a local option structwm42018-05-242-3/+29
| | | | Instead of accessing MPOpts.
* build: make encoding mode non-optionalwm42018-05-031-2/+0
| | | | Makes it easier to not break the build by confusing the ifdeffery.
* encode: get rid of the output packet queuewm42018-05-035-3/+33
| | | | | | | | | | | | Until recently, ao_lavc and vo_lavc started encoding whenever the core happened to send them data. Since audio and video are not initialized at the same time, and the muxer was not necessarily opened when the first encoder started to produce data, the resulting packets were put into a queue. As soon as the muxer was opened, the queue was flushed. Change this to make the core wait with sending data until all encoders are initialized. This has the advantage that we don't need to queue up the packets.
* encode: remove old timestamp handlingwm42018-05-031-46/+6
| | | | | This effectively makes --ocopyts the default. The --ocopyts option itself is also removed, because it's redundant.
* encode: rewrite half of itwm42018-04-291-185/+55
| | | | | | | | | | | | | The main change is that we wait with opening the muxer ("writing headers") until we have data from all streams. This fixes race conditions at init due to broken assumptions in the old code. This also changes a lot of other stuff. I found and fixed a few API violations (often things for which better mechanisms were invented, and the old ones are not valid anymore). I try to get away from the public mutex and shared fields in encode_lavc_context. For now it's still needed for some timestamp-related fields, but most are gone. It also removes some bad code duplication between audio and video paths.
* encode: cosmeticswm42018-04-201-25/+29
| | | | Mostly whitespace changes; some semantic preserving transformations.
* ao_alsa: actually report underruns to userwm42018-04-151-5/+5
| | | | | | | | | Print them as a warning. Note that there may be some cases where it underruns, without being a bad condition. This could possibly happen e.g. if the last chunk is written, and then it resumes playback some time after that. Eventually I want to add more code to avoid such spurious warnings.
* ao_pulse: reduce requested device buffer sizewm42018-04-151-1/+1
| | | | | | Same deal as with the previous commit for ALSA. Untested.
* ao_alsa: reduce requested buffer sizewm42018-04-151-2/+2
| | | | | | There is a dedicated thread for feeding audio to the ALSA API from a buffer with a larger size. There is little reason to have such a large device buffer.
* ao_alsa: add options for controlling period/buffer sizewm42018-04-151-8/+16
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* ao_openal: document the muted↔gain conversionJan Ekström2018-04-151-0/+3
| | | | This struck me as odd for a moment, so adding a comment.
* ao/openal: Add option to set buffering characteristicsLAGonauta2018-04-151-23/+62
| | | | | | | | | One can now set the number of buffers and the buffer size. This can reduce the CPU usage and the total latency stays mostly the same. As there are sync mechanisms the A/V sync continue intact and working. It also modifies 6.1 channel order, as per OpenAL spec and add AOPLAY_FINAL_CHUNK support
* ao/openal: Add better sample format and channel layout selectionLAGonauta2018-04-151-139/+73
| | | | Also re-added floating-point support.
* ao/openal: Add OpenAL Soft extension to get the correct latencyLAGonauta2018-04-151-1/+16
| | | | | | | | OpenAL Soft's AL_SOFT_source_latency extension allows one to correctly get the device output latency, facilitating the syncronization with video. Also added a simpler generic fallback that does not take into account latency of the device.
* ao/openal: Add support for direct channels outputLAGonauta2018-04-151-0/+10
| | | | | | | Uses OpenAL Soft's AL_DIRECT_CHANNELS_SOFT extension and can be controlled through a new CLI option, --openal-direct-channels. This allows one to send the audio data direrctly to the desired channel without effects applied.
* ao/openal: Add hardware mute supportLAGonauta2018-04-151-0/+12
| | | | | While the volume is set on the listener, mute is set on the sound source. Seemed easier that way.
* ao/openal: Use only one source for audio outputLAGonauta2018-04-151-52/+153
| | | | Floating point audio not supported on this commit.
* ao_opensles: let cfg_frames_per_buffer accept buffer size up to 0.5s at 192kHzTom Yan2018-04-051-1/+1
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* ao_opensles: remove useless cfg_sample_rateTom Yan2018-04-051-5/+0
| | | | We should always use the ao-neutral --audio-samplerate option.
* ao_opensles: bump device buffer size to 250msTom Yan2018-04-051-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Although half (non-fast track on sink rate) or one-third (non-fast track not on sink rate) of the buffer size of the created AudioTrack instance as the SL Enqueue buffer size is basically enough for dropout-free playback, only using the full size can avoid stutter upon (re)start of playback. Here are the various buffer sizes on different track/sink rate when on Bluetooth audio on Android O: aptX @ 48kHz: Sink rate: 48000 Hz 44100 Hz: 10632 frames (241.09 ms) 48000 Hz: 11544 frames (240.50 ms) 88200 Hz: 21216 frames (240.54 ms) 96000 Hz: 23088 frames (240.50 ms) 176400 Hz: 42384 frames (240.27 ms) 192000 Hz: 46128 frames (240.25 ms) SBC/AAC/aptX @ 44.1kHz: Sink rate: 44100 Hz 44100 Hz: 10776 frames (244.35 ms) 48000 Hz: 11748 frames (244.75 ms) 88200 Hz: 21552 frames (244.35 ms) 96000 Hz: 23448 frames (244.25 ms) 176400 Hz: 43056 frames (244.08 ms) 192000 Hz: 46848 frames (244.00 ms) The above results were produced with the following code: import android.media.AudioAttributes; import android.media.AudioFormat; import android.media.AudioTrack; class AudioInfo { public static void main(String[] args) { int nosr = AudioTrack.getNativeOutputSampleRate(3); System.out.printf("Sink rate: %d Hz\n", nosr); int[] rates = {44100,48000,88200,96000,176400,192000}; for (int rate: rates) { AudioAttributes aa = new AudioAttributes.Builder().setFlags(256).build(); AudioFormat af = new AudioFormat.Builder().setSampleRate(rate).build(); AudioTrack at = new AudioTrack(aa, af, 4, 1, 0); int sr = at.getSampleRate(); int bs = at.getBufferSizeInFrames(); float ms = bs * (float) 1000 / sr; at.release(); System.out.printf("%d Hz: %d frames (%.2f ms)\n", sr, bs, ms); } } } Therefore bumping the device buffer size to 250ms.
* ao_opensles: do one buffer onlyTom Yan2018-04-051-15/+8
| | | | Doing two buffers causes stutters upon (re)start of playback on Android O for all kinds of sinks.
* ao_opensles: re-flow interface/configuration retrievalJan Ekström2018-03-241-9/+18
| | | | | This manages to make the code more readable. Thanks to MakeGho@IRCnet for the snippet on which this was based.
* ao_opensles: fix audio sync using device latency extensionAman Gupta2018-03-231-3/+20
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* ao_sdl: fix default buffer sizewm42018-03-081-5/+6
| | | | | | | | If you set desired.samples to 0, SDL will return a default buffer size on obtained.samples. This was broken, because ceil_power_of_two(0) returns 1. Since 0 is usually not considered a power of two, this is probably correct, but we still want to set desired.samples to 0 in this case.
* ao: do not allow actual buffer size of 0wm42018-03-081-0/+1
| | | | | | | | | You can use --audio-buffer=0 to minimize the audio buffer size. But if the AO reports no device buffer size (like e.g. ao_jack does), then the buffer size is actually 0, and playback can never work properly. Make it fallback to a size of 1, which is unlikely to work properly, but you get what you asked for, instead of a freeze.
* ao_opensles: bump device buffer size to 200mstomty892018-03-071-1/+1
| | | While the soft buffer size is already by default 200ms, it is not enough to guarantee dropout-free playback on Bluetooth audio. Bumping the device buffer size to the same value seems to suffice.
* ao_opensles: remove set_play_state()tomty892018-03-071-10/+1
| | | Set play state to playing in init() instead. We no longer touch the play state afterwards.
* ao_opensles: clear buffer queue in reset()tomty892018-03-071-1/+2
| | | Avoid resume() from causing SL_RESULT_BUFFER_INSUFFICIENT ("Failed to Enqueue: 7" when seek or resume from pause).
* build: drop support for SDL1wm42018-02-131-1/+1
| | | | | For some reason it was supported for ao_sdl because we've only used SDL1 API.
* ao_null: add --ao-null-format option for debuggingwm42018-01-301-0/+5
| | | | Helpful especially to test spdif fallback and so on.
* ao_alsa: reduce verbosity at -vwm42018-01-251-10/+10
| | | | | | | Always make the hw params dump function use MSGL_DEBUG, and remove the MSGL_V use. That means you need -v -v to see them. The detailed information is usually not very interesting, so this reduces the log noise.
* audio: fix annyoing af_get_best_sample_formats() definitionwm42018-01-256-6/+6
| | | | | | | | | | | | | | | | The af_get_best_sample_formats() function had an argument of int[AF_FORMAT_COUNT], which is slightly incorrect, because it's 0 terminated and should in theory have AF_FORMAT_COUNT+1 entries. It won't actually write this many formats (since some formats are fundamentally incompatible), but it still feels annoying and incorrect. So fix it, and require that callers pass an AF_FORMAT_COUNT+1 array. Note that the array size has no meaning in C function arguments (just another issue with C static arrays being weird and stupid), so get rid of it completely. Not changing the af_lavcac3enc use, since that is rewritten in another branch anyway.
* Fix undefined preprocessor behaviorwm42018-01-182-3/+10
| | | | | | | | | | | This commit eliminates the following clang warning: warning: macro expansion producing 'defined' has undefined behavior [-Wexpansion-to-defined] Going by the clang commit message, this seems to be explicitly specified as UB by the standard, and they added this warning because MSVC apparently results in different behavior. Whatever, we can just avoid the warning with some small changes.
* Fix various typos in log messagesNicolas F2017-12-031-1/+1
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* ao: minor simplification to gain processing codewm42017-11-301-4/+3
| | | | | Cosmetic move of a variable, and consider an adjustment below 1/256 or so not worth applying (even in the float case).
* ao: simplify hack for float atomicswm42017-11-302-16/+4
| | | | | | | stdatomic.h defines no atomic_float typedef. We can't just use _Atomic unconditionally, because we support compilers without C11 atomics. So just create a custom atomic_float typedef in the wrapper, which uses _Atomic in the C11 code path.
* audio: add audio softvol processing to AOwm42017-11-295-0/+75
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
* ao_alsa: change license to LGPLwm42017-11-231-27/+15
| | | | | | | | | | | | | | | | | | | | | | | | Looks like this is covered by LGPL relicensing agreements now. Notes about contributors who could not be reached or who didn't agree: Commit 7fccb6486e has tons of mp_msg changes look like they are not copyrightable (even if they were, all mp_msg calls were rewritten in mpv times again). The additional play() change looks suspicious, but the function was rewritten several times anyway (first time after that commit in 4f40ec312). Commit 89ed1748ae was rewritten in commit 325311af3 and then again several times after that. Basically all this code is unnecessary in modern mpv and has been removed. No code survived from the following commits: 4d31c3c53, 61ecf838f2, d38968bd, 4deb67c3f. At least two cosmetic typo fixes are not considered as well. Commit 22bb046ad is reverted (this wasn't a valid warning anyway, just a C++-ism icc applied to C). Using the constants is nicer, but at least I don't have to decide whether that change was copyrightable.
* ao_alsa: don't convert twice on retrywm42017-11-231-2/+1
| | | | Obscure corner case.
* audio/out: initialize an array to avoid confusing static analyzerwm42017-10-271-1/+1
| | | | | | | I _think_ this confuses Coverity and it thinks there is uninitialized data to be read. Initialize the array to change/remove the warning, or if there's a real problem, to make it easier to detect. (Basically apply defensive coding.)
* Add checks for HAVE_GPL to various GPL-only source fileswm42017-10-103-0/+12
| | | | | | | | This should actually cover all of them, if you take into account that some unchanged GPL source files include header files with such checks. Also this was done already for the libaf derived code. This is only for "safety" and to avoid misunderstandings.
* command: drop "audio-out-detected-device" propertywm42017-10-093-11/+0
| | | | | | Coreaudio stopped setting it a few releases ago (66a958bb4fa). There is not much of a user- or API-visible change, so remove it without deprecation.
* audio_buffer: remove dependency on mp_audiowm42017-09-211-46/+43
| | | | | | | | | Just reimplement it in some way, as mp_audio is GPL-only. Actually I wanted to get rid of audio_buffer.c completely (and instead have a list of mp_aframes), but to do so would require rewriting some more player core audio code. So to get this LGPL relicensing over quickly, just do some extra work.
* ao_rsound: allow setting the hostwm42017-08-211-0/+3
| | | | | | | | Completely untested (rsound dev libs unavailable on my system). Trivial enough that it's very likely that it'll just work. No port selection, but could be added by parsing it as part of the device name. Should fix #4714.
* audio: introduce a new type to hold audio frameswm42017-08-162-8/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is pretty pointless, but I believe it allows us to claim that the new code is not affected by the copyright of the old code. This is needed, because the original mp_audio struct was written by someone who has disagreed with LGPL relicensing (it was called af_data at the time, and was defined in af.h). The "GPL'ed" struct contents that surive are pretty trivial: just the data pointer, and some metadata like the format, samplerate, etc. - but at least in this case, any new code would be extremely similar anyway, and I'm not really sure whether it's OK to claim different copyright. So what we do is we just use AVFrame (which of course is LGPL with 100% certainty), and add some accessors around it to adapt it to mpv conventions. Also, this gets rid of some annoying conventions of mp_audio, like the struct fields that require using an accessor to write to them anyway. For the most part, this change is only dumb replacements of mp_audio related functions and fields. One minor actual change is that you can't allocate the new type on the stack anymore. Some code still uses mp_audio. All audio filter code will be deleted, so it makes no sense to convert this code. (Audio filters which are LGPL and which we keep will have to be ported to a new filter infrastructure anyway.) player/audio.c uses it because it interacts with the old filter code. push.c has some complex use of mp_audio and mp_audio_buffer, but this and pull.c will most likely be rewritten to do something else.
* ao_wasapi: remove old commentKevin Mitchell2017-08-071-1/+1
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* ao_wasapi: reorganize wasapi.hKevin Mitchell2017-08-072-12/+8
| | | | | Remove dead declarations. Move macro only used in wasapi_utils.c closer to use. Rearrange declaration order.
* ao_wasapi: deduplicate wasapi sample format selectionKevin Mitchell2017-08-071-49/+40
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* ao_wasapi: clean up find_formats logicKevin Mitchell2017-08-071-75/+45
| | | | | There were too many functions within functions, too much going on in if clauses and duplicated code. Fix it.
* ao_wasapi: return bool instead of HRESULT from thread_initKevin Mitchell2017-08-073-38/+48
| | | | | | | | Any bad HRESULTs should have been printed already and lots of failure modes don't have an HRESULT leading to awkward hr = E_FAIL business. This also checks the exit status of GetBufferSize in the align hack. A final fatal message is added if either of the retry hacks fail.
* ao_oss: fix a dumb calculationwm42017-07-211-2/+3
| | | | | | | period_size used the wrong unit, and even if the unit had been correct, was assigned the wrong value. Probably fixes #4642.
* ao_wasapi: remove redundant / outdated commentKevin Mitchell2017-07-101-1/+0
| | | | | Where this was moved from, it made slightly more sense. Here what the comment is trying to say is already pretty obvious from the code.
* ao_waspi: use switch for handling fix_format errorsKevin Mitchell2017-07-101-5/+8
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* ao_wasapi: don't repeat format negotiation on align hackKevin Mitchell2017-07-101-4/+7
| | | | | Even if it did return a different result, the bufferFrameCount from the align hack would be wrong anyway.
* ao_wasapi: fix leak on align hackKevin Mitchell2017-07-101-1/+1
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* audio/out: fix comment typoKevin Mitchell2017-07-091-1/+1
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* ao_wasapi: enable packed 24 bit outputKevin Mitchell2017-07-091-1/+1
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* audio/out: correct copy length in ao_read_data_convertedKevin Mitchell2017-07-091-6/+7
| | | | | | | | | | | | | Previously, the entire convert_buffer was being copied to the desination without regard to the fact that it may be packed and therefore smaller. The allocated conversion buffer was also way to big bytes * (channels * samples) ** 2 instead of bytes * channels * samples
* ao_wasapi: reorder channels and samplerates to speed up searchKevin Mitchell2017-07-091-4/+4
| | | | | | | This shouldn't affect which are chosen, but it should speed up the search by putting more common configurations earlier so that a working sample format and sample rates can be found sooner obviating the need to search them for each iteration of the outer loops.
* ao_wasapi: minor cosmetic fixesKevin Mitchell2017-07-091-3/+3
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* ao_wasapi: try correct initial formatKevin Mitchell2017-07-091-1/+4
| | | | | | | | | The loop to select the native wasapi_format for the incoming audio was not breaking correctly when it found the most desirable format. It therefore executed completely leaving the