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* audio: remove unreferenced af_lavrresamplewm42019-09-191-112/+0
| | | | | | | | | | | | | | This filter wasn't referenced anywhere and thus was dead code. It should have been in the audio filter list in user_filters.c. This was intended as compatibility wrapper (to avoid breaking old command lines and config files), and has no real use. Apparently I forgot to add it to the filter list (did I even test this shit?), and so it was rotting around for 1.5 years doing nothing (just like myself). Note that users can just use the libavfilter provided filter to force resampling, just that it has a different name and different options. There's also af_format to force inserting auto conversion through the internal f_swsresample filter.
* audio_buffer: fix some more theoretical UBwm42019-09-191-0/+3
| | | | | | | This may call memmove() with size==0 and a NULL data pointer. In addition to this being UB with memmove(), I think it's UB to do arithmetic on a NULL pointer too. Of course, this doesn't matter in practice at all, and is just stupidity to torture programmers.
* ad_lavc: skip fully skipped frameswm42019-09-191-1/+5
| | | | | | | | | | | Fixes stupid messages with a opus/mkv test file that had an absurdly huge codec delay. This file fully skips several frames at the start. ad_lavc.c trimmed these frames to 0 samples and returned them. The next layer (f_decoder_wrapper.c) saw discontinuous PTS values, because the PTS values increased by a frame, but amounted to 0 audio samples. This was harmless, but logged PTS discontinuity errors.
* Implement backwards playbackwm42019-09-192-0/+54
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | See manpage additions. This is a huge hack. You can bet there are shit tons of bugs. It's literally forcing square pegs into round holes. Hopefully, the manpage wall of text makes it clear enough that the whole shit can easily crash and burn. (Although it shouldn't literally crash. That would be a bug. It possibly _could_ start a fire by entering some sort of endless loop, not a literal one, just something where it tries to do work without making progress.) (Some obvious bugs I simply ignored for this initial version, but there's a number of potential bugs I can't even imagine. Normal playback should remain completely unaffected, though.) How this works is also described in the manpage. Basically, we demux in reverse, then we decode in reverse, then we render in reverse. The decoding part is the simplest: just reorder the decoder output. This weirdly integrates with the timeline/ordered chapter code, which also has special requirements on feeding the packets to the decoder in a non-straightforward way (it doesn't conflict, although a bugmessmass breaks correct slicing of segments, so EDL/ordered chapter playback is broken in backward direction). Backward demuxing is pretty involved. In theory, it could be much easier: simply iterating the usual demuxer output backward. But this just doesn't fit into our code, so there's a cthulhu nightmare of shit. To be specific, each stream (audio, video) is reversed separately. At least this means we can do backward playback within cached content (for example, you could play backwards in a live stream; on that note, it disables prefetching, which would lead to losing new live video, but this could be avoided). The fuckmess also meant that I didn't bother trying to support subtitles. Subtitles are a problem because they're "sparse" streams. They need to be "passively" demuxed: you don't try to read a subtitle packet, you demux audio and video, and then look whether there was a subtitle packet. This means to get subtitles for a time range, you need to know that you demuxed video and audio over this range, which becomes pretty messy when you demux audio and video backwards separately. Backward display is the most weird (and potentially buggy) part. To avoid that we need to touch a LOT of timing code, we negate all timestamps. The basic idea is that due to the navigation, all comparisons and subtractions of timestamps keep working, and you don't need to touch every single of them to "reverse" them. E.g.: bool before = pts_a < pts_b; would need to be: bool before = forward ? pts_a < pts_b : pts_a > pts_b; or: bool before = pts_a * dir < pts_b * dir; or if you, as it's implemented now, just do this after decoding: pts_a *= dir; pts_b *= dir; and then in the normal timing/renderer code: bool before = pts_a < pts_b; Consequently, we don't need many changes in the latter code. But some assumptions inhererently true for forward playback may have been broken anyway. What is mainly needed is fixing places where values are passed between positive and negative "domains". For example, seeking and timestamp user display always uses positive timestamps. The main mess is that it's not obvious which domain a given variable should or does use. Well, in my tests with a single file, it suddenly started to work when I did this. I'm honestly surprised that it did, and that I didn't have to change a single line in the timing code past decoder (just something minor to make external/cached text subtitles display). I committed it immediately while avoiding thinking about it. But there really likely are subtle problems of all sorts. As far as I'm aware, gstreamer also supports backward playback. When I looked at this years ago, I couldn't find a way to actually try this, and I didn't revisit it now. Back then I also read talk slides from the person who implemented it, and I'm not sure if and which ideas I might have taken from it. It's possible that the timestamp reversal is inspired by it, but I didn't check. (I think it claimed that it could avoid large changes by changing a sign?) VapourSynth has some sort of reverse function, which provides a backward view on a video. The function itself is trivial to implement, as VapourSynth aims to provide random access to video by frame numbers (so you just request decreasing frame numbers). From what I remember, it wasn't exactly fluid, but it worked. It's implemented by creating an index, and seeking to the target on demand, and a bunch of caching. mpv could use it, but it would either require using VapourSynth as demuxer and decoder for everything, or replacing the current file every time something is supposed to be played backwards. FFmpeg's libavfilter has reversal filters for audio and video. These require buffering the entire media data of the file, and don't really fit into mpv's architecture. It could be used by playing a libavfilter graph that also demuxes, but that's like VapourSynth but worse.
* ao_opensles: fix delayed audiosfan52019-09-021-1/+1
| | | | | This was forgotten in commit 5a8c48fde2a26fe00c3552e3ccf83a965b6d3576 when the number of buffers was reduced to 1.
* ao/audiounit: include AVAudioSession buffer in latency calcAman Gupta2019-04-051-1/+1
| | | | Signed-off-by: Aman Gupta <aman@tmm1.net>
* ao/audiounit: improve a/v syncAman Gupta2019-04-053-4/+18
| | | | | | | This more closely mimics ao_coreaudio, on which this driver was originally based. Signed-off-by: Aman Gupta <aman@tmm1.net>
* Merge commit '559a400ac36e75a8d73ba263fd7fa6736df1c2da' into ↵Anton Kindestam2018-12-053-5/+33
|\ | | | | | | | | | | wm4-commits--merge-edition This bumps libmpv version to 1.103
| * player: get rid of mpv_global.optswm42018-05-241-2/+4
| | | | | | | | | | | | | | | | This was always a legacy thing. Remove it by applying an orgy of mp_get_config_group() calls, and sometimes m_config_cache_alloc() or mp_read_option_raw(). win32 changes untested.
| * ao: use a local option structwm42018-05-242-3/+29
| | | | | | | | Instead of accessing MPOpts.
* | ad_spdif: cosmetic alignmentJan Ekström2018-10-301-1/+1
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* | ad_spdif: fix DTS-HD HRA handlingJan Ekström2018-10-301-2/+10
| | | | | | | | | | | | | | | | | | | | | | | | Apparently, for bit streaming DTS-HD MA is specified to be handled as an eight channel (7.1) bit stream, while DTS-HD HRA is specified to be handled as a stereo bit stream. Define a variable for this, and utilize it to set the correct values for both the DTS-HD bit streaming rate, as well as the channel count for the SPDIF encoder. Fixes #6148
* | ao_audiounit: rename pause function to resetJosh Lehman2018-09-301-1/+1
| | | | | | | | | | AudioUnit output driver uses the pull based api so it should have a reset function instead of a pause function.
* | ao_alsa: log the ALSA state if we get a non-XRUN errorJan Ekström2018-09-291-2/+4
| | | | | | | | | | The ALSA state generally can tell us more information in case we get an unexpected error.
* | ao_alsa: handle XRUNs separately from other errorsJan Ekström2018-09-291-2/+7
| | | | | | | | | | | | | | | | | | According to ALSA doxy, EPIPE is a synonym to SND_PCM_STATE_XRUN, and that is a state that we should attempt to automatically recover from. In case recovery fails, log an error and return zero. A warning message will still be output for each XRUN since those are not something we should generally be receiving.
* | ao_alsa: early exit get_space if paused or ALSA is not readyJan Ekström2018-09-291-0/+5
| | | | | | | | | | | | | | | | | | | | | | This has been way too long coming, and for me to notice that a whole lot of ao_alsa functions do an early return if the AO is paused. For the STATE_SETUP case, I had this reproduced once, and never since. Still, seems like we can start calling this function before the ALSA device has been fully initialized so we might as well early exit in that case.
* | ao_jack: only auto-connect to audio portsNiklas Haas2018-09-261-1/+2
| | | | | | | | | | This prevents ao_jack from auto-connecting to MIDI ports (or other, hypothetical future port types).
* | ao_pulse: fix tlength calculationTom Yan2018-09-013-17/+3
| | | | | | | | also remove the now unused non-sensical af_fmt_seconds_to_bytes.
* | Revert "ao_openal: enable building on OSX"Michael Hoang2018-08-261-14/+0
| | | | | | | | | | | | This reverts commit af6126adbe61fb2b6cc780025246d33df93072e6. Apple's OpenAL support is ridiculously out of date, revert back to just using OpenAL Soft on macOS (fixes #4645).
* | af_rubberband: reset delay to 0 on resetHector Martin2018-08-251-0/+2
| | | | | | | | This fixes A-V drift on seeking
* | ao_opensles: set numBuffers to 8Tom Yan2018-08-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Apparently some Android builds/forks require this for Bluetooth audio to work as they unexpectedly accept fast flag for it. Shouldn't cause any side-effect (e.g. buffer requirement increased when on wired audio). It's a hardcoded default in the upstream AAudio implementation anyway. Ref.: https://android.googlesource.com/platform/frameworks/av/+/android-8.0.0_r1/media/libaaudio/src/legacy/AudioStreamTrack.cpp#109 https://android.googlesource.com/platform/frameworks/wilhelm/+/android-8.0.0_r1/src/android/AudioPlayer_to_android.cpp#1680 https://android.googlesource.com/platform/frameworks/av/+/android-8.0.0_r1/media/libaudioclient/AudioTrack.cpp#488
* | audio/format: decouple af_fmt_is_planar from af_fmt_to_planarTom Yan2018-08-111-16/+15
| | | | | | | | | | so that af_fmt_to_planar (and hence af_fmt_from_planar) can just return the input when it is not an interleaved (planar) format.
* | ao_opensles: rework the heuristic of buffer/enqueue size settingTom Yan2018-08-051-18/+36
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ao->device_buffer will only affect the enqueue size if the latter is not specified. In other word, its intended purpose will solely be setting/guarding the soft buffer size. This guarantees that the soft buffer size will be consistent no matter a specific enqueue size is set or not. (In the past it would drop to the default of the generic audio-buffer option.) opensles-frames-per-buffer has been renamed to opensles-frames-per -enqueue, as it was never purposed to set the soft buffer size. It will only make sure the size is never smaller than itself, just as before. opensles-buffer-size-in-ms is introduced to allow easy tuning of the relative (i.e. in time) soft buffer size (and enqueue size, unless the aforementioned option is set). As "device buffer" never really made sense in this AO, this option OVERRIDES audio-buffer whenever its value (including the default) is larger than 0. Setting opensl-buffer-size-in-ms to 1 allows you to equate the soft buffer size to the absolute enqueue size set with opensl-frames-per -enqueue conveniently (unless it is less than 1ms). When both are set to 0, audio-buffer will be the ultimate fallback. If audio-buffer is also 0, the AO errors out.
* | ao_opensles: allow s32 and float outputTom Yan2018-08-051-27/+15
| | | | | | | | | | OpenSLES (and its AudioTrack backend) in Android can take 32-bit fixed and floating point input since Android L (API 21).
* | audio/format: minor fix for af_fmt_from_planarTom Yan2018-08-051-2/+4
| | | | | | | | See af_fmt_to_planar.
* | ao_alsa: simplify get_space()Jan Ekström2018-06-041-6/+10
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* | ao_alsa: replace snd_pcm_status() with snd_pcm_avail() in get_space()Muhammad Faiz2018-06-041-5/+4
|/ | | | | | | | | | Fixes a bug with alsa dmix on Fedora 29. After several minutes, audio suddenly becomes bad and muted. Actually, I don't know what causes this. Probably this is a bug in alsa. In any case, as snd_pcm_status() returns not only 'avail', but also other fields such as tstamp, htstamp, etc, this could be considered a good simplification, as only avail is required for this function.
* 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.
*