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* audio/filter: use new option APIwm42013-07-221-12/+5
| | | | | | | | | | | | | Make the VF/VO/AO option parser available to audio filters. No audio filter uses this yet, but it's still a quite intrusive change. In particular, the commands for manipulating filters at runtime completely change. We delete the old code, and use the same infrastructure as for video filters. (This forces complete reinitialization of the filter chain, which hopefully isn't a problem for any use cases. The old code forced reinitialization too, but it could potentially allow a filter to cache things; e.g. consider loaded ladspa plugins and such.)
* audio/decode: remove macro crapwm42013-07-221-5/+18
| | | | | Declare decoders directly, instead of using the LIBAD_EXTERN macro. This is simpler (no weird magic) and more extensible.
* mplayer: fix incorrect audio sync after format changeswm42013-07-111-8/+2
| | | | | | | | | | | This is not directly related to the handling of format changes itself, but playing audio normally after the change. This was broken: the output byte rate was not recalculated, so audio-video sync was simply broken. Fix this by calculating the byte rate on the fly, instead of storing it in sh_audio. Format changes are relatively common (switches between stereo and 5.1 in TV recordings), so this fixes a somewhat critical bug.
* demux: remove facility for partial packet readswm42013-07-111-11/+0
| | | | | | | | | | | | | | | | | | Partial packet reads were needed because the video/audio parsers were working on top of them. So it could happen that a parser read a part of a packet, and returned that to the decoder. With libavformat/libavcodec, packets are already parsed, and everything is much simpler. Most of the simplifications in ad_spdif could have been done earlier. Remove some other stuff as well, like the questionable slave mode start time reporting (could be replaced by proper code, but we don't bother). Remove the unused skip_audio_frame() functionality as well (it was used by old demuxers). Some functions become private to demux.c, like demux_fill_buffer(). Introduce new packet read functions, which have simpler semantics. Packets returned from them are owned by the caller, and all packets in the demux.c packet queue are considered unread. Remove special code that dropped subtitle packets with size 0. This used to be needed because it caused special cases in the old code.
* audio: remove decoder input bufferwm42013-07-101-11/+0
| | | | This was unused.
* Remove old demuxerswm42013-07-071-0/+2
| | | | | | | | | | Delete demux_avi, demux_asf, demux_mpg, demux_ts. libavformat does better than them (except in rare corner cases), and the demuxers have a bad influence on the rest of the code. Often they don't output proper packets, and require additional audio and video parsing. Most work only in --no-correct-pts mode. Remove them to facilitate further cleanups.
* options: remove --stereowm42013-06-131-2/+0
| | | | | | | Whatever this was supposed to be originally, it doesn't have much value anymore. It just forced ad_mpg123 to upmix mono to stereo by default (the audio chain can do that). As an option, it was mostly useless and misleading, so get rid of it.
* audio: print channel map additionally to channel count on terminalwm42013-05-121-4/+5
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* core: use channel map on demuxer level toowm42013-05-121-9/+9
| | | | | | | | | | | | | | | This helps passing the channel layout correctly from decoder to audio filter chain. (Because that part "reuses" the demuxer level codec parameters, which is very disgusting.) Note that ffmpeg stuff already passed the channel layout via mp_copy_lav_codec_headers(). So other than easier dealing with the demuxer/decoder parameters mess, there's no real advantage to doing this. Make the --channels option accept a channel map. Since simple numbers map to standard layouts with the given number of channels, this is downwards compatible. Likewise for demux_rawaudio.
* audio/out: switch to channel mapwm42013-05-121-3/+4
| | | | | | This actually breaks audio for 5/6/8 channels. There's no reordering done yet. The actual reordering will be done inside of af_lavrresample and has to be made part of the format negotiation.
* audio: add some setters for mp_audio, and require filters to use themwm42013-05-121-11/+9
| | | | | | | | | | | | | | | | mp_audio has some redundant fields. Setters like mp_audio_set_format() initialize these properly. Also move the mp_audio struct to a the file audio.c. We can remove a mysterious line of code from af.c: in.format |= af_bits2fmt(in.bps * 8); I'm not sure if this was ever actually needed, or if it was some kind of "make it work" quick-fix that works against the way things were supposed to work. All filters etc. now set the format correctly, so if there ever was a need for this code, it's definitely gone.
* af: streamline format negotiationwm42013-04-131-6/+4
| | | | | | | | | | | | | Add dummy input and output filters to remove special cases in the format negotiation code (af_fix_format_conversion() etc.). The output of the filter chain is now negotiated in exactly the same way as normal filters. Negotiate setting the sample rate in the same way as other audio parameters. As a side effect, the resampler is inserted at the start of the filter chain instead of the end, but that shouldn't matter much, especially since conversion and channel mixing are conflated into the same filter (due to libavresample's API).
* options: remove --af-advwm42013-04-131-1/+1
| | | | | | | Anything this option did has been removed in the preceding 3 commits. Note that even though these options sounded like a good idea (like setting accuracy vs. speed tradeoffs), they were not really properly implemented.
* audio: remove float processing optionwm42013-04-131-10/+2
| | | | | | | | | | | | | | The only thing this option did was changing the behavior of af_volume. The option decided what sample format af_volume would use, but only if the sample format was not already float. If the option was set, it would default to float, otherwise to S16. Remove use of the option and all associated code, and make af_volume always use float (unless a af_volume specific sub-option is set). Silence maximum value tracking. This message is printed when the filter is destroyed, and it's slightly annoying. Was enabled due to enabling float by default.
* core: redo how codecs are mapped, remove codecs.confwm42013-02-101-151/+77
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Use codec names instead of FourCCs to identify codecs. Rewrite how codecs are selected and initialized. Now each decoder exports a list of decoders (and the codec it supports) via add_decoders(). The order matters, and the first decoder for a given decoder is preferred over the other decoders. E.g. all ad_mpg123 decoders are preferred over ad_lavc, because it comes first in the mpcodecs_ad_drivers array. Likewise, decoders within ad_lavc that are enumerated first by libavcodec (using av_codec_next()) are preferred. (This is actually critical to select h264 software decoding by default instead of vdpau. libavcodec and ffmpeg/avconv use the same method to select decoders by default, so we hope this is sane.) The codec names follow libavcodec's codec names as defined by AVCodecDescriptor.name (see libavcodec/codec_desc.c). Some decoders have names different from the canonical codec name. The AVCodecDescriptor API is relatively new, so we need a compatibility layer for older libavcodec versions for codec names that are referenced internally, and which are different from the decoder name. (Add a configure check for that, because checking versions is getting way too messy.) demux/codec_tags.c is generated from the former codecs.conf (minus "special" decoders like vdpau, and excluding the mappings that are the same as the mappings libavformat's exported RIFF tables). It contains all the mappings from FourCCs to codec name. This is needed for demux_mkv, demux_mpg, demux_avi and demux_asf. demux_lavf will set the codec as determined by libavformat, while the other demuxers have to do this on their own, using the mp_set_audio/video_codec_from_tag() functions. Note that the sh_audio/video->format members don't uniquely identify the codec anymore, and sh->codec takes over this role. Replace the --ac/--vc/--afm/--vfm with new --vd/--ad options, which provide cover the functionality of the removed switched. Note: there's no CODECS_FLAG_FLIP flag anymore. This means some obscure container/video combinations (e.g. the sample Film_200_zygo_pro.mov) are played flipped. ffplay/avplay doesn't handle this properly either, so we don't care and blame ffmeg/libav instead.
* dec_audio: uncrustifywm42013-02-091-192/+201
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* audio: improve decoder open failure handlingUoti Urpala2012-12-031-0/+2
| | | | | | | | | | | | | | | | Reinitialize sh_audio->samplesize and sample_format before falling back to another audio decoder (some decoders rely on default values). Remove code setting these fields from demux_mkv and demux_lavf (no decoder should depend on demuxer-set values for these fields). Conflicts: audio/decode/ad_lavc.c Merged from mplayer2 commit 6b9567. The changes to ad_lavc.c are not merged, as they are very specific to the mplayer2 libavresample hack; we deplanarize manually, so we can't get unsupported sample formats yet (except on raw audio with "pcm_f64le", as we don't support AV_SAMPLE_FMT_DBL in the audio chain).
* core: fix crash when video filter returns inf as PTSwm42012-11-201-1/+1
| | | | | | | | | | | | | | | | | | | When a video filter returned inf as PTS, the player crashed. One reason for this was that decode_audio() was called with a negative minlen parameter, which at some point caused it to call a memory allocation function with a ridiculous value, triggering an out of memory code path in talloc.c. (talloc.c has been modified to abort() on out of memory situations.) Fix this by sanity checking minlen in decode_audio(). (The check against outbuf->len always succeeded, because it's an unsigned comparison.) Make an existing sanity check in mplayer.c more robust: check for NaN too, which happens if the video PTS is inf. This happened with "-vf pullup,softpulldown" (but is not triggered when the following commit is applied).
* Rename directories, move files (step 2 of 2)wm42012-11-121-7/+7
| | | | | | | | | | | | Finish renaming directories and moving files. Adjust all include statements to make the previous commit compile. The two commits are separate, because git is bad at tracking renames and content changes at the same time. Also take this as an opportunity to remove the separation between "common" and "mplayer" sources in the Makefile. ("common" used to be shared between mplayer and mencoder.)
* Rename directories, move files (step 1 of 2) (does not compile)wm42012-11-121-0/+462
Tis drops the silly lib prefixes, and attempts to organize the tree in a more logical way. Make the top-level directory less cluttered as well. Renames the following directories: libaf -> audio/filter libao2 -> audio/out libvo -> video/out libmpdemux -> demux Split libmpcodecs: vf* -> video/filter vd*, dec_video.* -> video/decode mp_image*, img_format*, ... -> video/ ad*, dec_audio.* -> audio/decode libaf/format.* is moved to audio/ - this is similar to how mp_image.* is located in video/. Move most top-level .c/.h files to core. (talloc.c/.h is left on top- level, because it's external.) Park some of the more annoying files in compat/. Some of these are relicts from the time mplayer used ffmpeg internals. sub/ is not split, because it's too much of a mess (subtitle code is mixed with OSD display and rendering). Maybe the organization of core is not ideal: it mixes playback core (like mplayer.c) and utility helpers (like bstr.c/h). Should the need arise, the playback core will be moved somewhere else, while core contains all helper and common code.