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* Merge branch 'planar_audio'wm42013-11-126-526/+331
|\ | | | | | | | | Conflicts: audio/out/ao_lavc.c
| * audio: add support for using non-interleaved audio from decoders directlywm42013-11-125-446/+316
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Most libavcodec decoders output non-interleaved audio. Add direct support for this, and remove the hack that repacked non-interleaved audio back to packed audio. Remove the minlen argument from the decoder callback. Instead of forcing every decoder to have its own decode loop to fill the buffer until minlen is reached, leave this to the caller. So if a decoder doesn't return enough data, it's simply called again. (In future, I even want to change it so that decoders don't read packets directly, but instead the caller has to pass packets to the decoders. This fits well with this change, because now the decoder callback typically decodes at most one packet.) ad_mpg123.c receives some heavy refactoring. The main problem is that it wanted to handle format changes when there was no data in the decode output buffer yet. This sounds reasonable, but actually it would write data into a buffer prepared for old data, since the caller doesn't know about the format change yet. (I.e. the best place for a format change would be _after_ writing the last sample to the output buffer.) It's possible that this code was not perfectly sane before this commit, and perhaps lost one frame of data after a format change, but I didn't confirm this. Trying to fix this, I ended up rewriting the decoding and also the probing.
| * ad_mpg123: reduce ifdefferywm42013-11-121-47/+2
| | | | | | | | Drop support for anything before 1.14.0.
| * dec_audio: fix behavior on format changeswm42013-11-121-3/+1
| | | | | | | | | | Decoder overwrites parameters in sh_audio, but we still have old audio in the old format to filter.
| * audio/filter: fix mul/delay scale and valueswm42013-11-121-4/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | Before this commit, the af_instance->mul/delay values were in bytes. Using bytes is confusing for non-interleaved audio, so switch mul to samples, and delay to seconds. For delay, seconds are more intuitive than bytes or samples, because it's used for the latency calculation. We also might want to replace the delay mechanism with real PTS tracking inside the filter chain some time in the future, and PTS will also require time-adjustments to be done in seconds. For most filters, we just remove the redundant mul=1 initialization. (Setting this used to be required, but not anymore.)
| * audio: switch output to mp_audio_bufferwm42013-11-122-39/+22
| | | | | | | | | | | | Replace the code that used a single buffer with mp_audio_buffer. This also enables non-interleaved output operation, although it's still disabled, and no AO supports it yet.
| * audio/filter: prepare filter chain for non-interleaved audiowm42013-11-121-6/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Based on earlier work by Stefano Pigozzi. There are 2 changes: 1. Instead of mp_audio.audio, mp_audio.planes[0] must be used. 2. mp_audio.len used to contain the size of the audio in bytes. Now mp_audio.samples must be used. (Where 1 sample is the smallest unit of audio that covers all channels.) Also, some filters need changes to reject non-interleaved formats properly. Nothing uses the non-interleaved features yet, but this is needed so that things don't just break when doing so.
* | demux_mkv: support some raw PCM variantswm42013-11-111-23/+11
|/ | | | | | | | | | This affects 64 bit floats and big endian integer PCM variants (basically crap nobody uses). Possibly not all MS-muxed files work, but I couldn't get or produce any samples. Remove a bunch of format tags that are not needed anymore. Most of these were used by demux_mov, which is long gone. Repurpose/abuse 'twos' as mpv-internal tag for dealing with the PCM variants mentioned above.
* ad_spdif: change API usage so that it works on Libavwm42013-11-101-3/+9
| | | | | | | | | | | Apparently we were using FFmpeg-specific APIs. I have no idea whether this code is correct on both FFmpeg and Libav (no examples, bad doxygen... why do they even complaint aht people are using their APIs incorrectly?), but it appears to work on FFmpeg. That was also the case before commit ebc4ccb though, where it used internal libavformat symbols. Untested on Libav, Travis will tell us.
* Remove sh_audio->samplesizewm42013-11-093-8/+3
| | | | | | | | | This member was redundant. sh_audio->sample_format indicates the sample size already. The TV code is a bit strange: the redundant sample size was part of the internal TV interface. Assume it's really redundant and not something else. The PCM decoder ignores the sample size anyway.
* ad_spdif: fix libavformat API usagewm42013-11-091-107/+76
| | | | | | | This accessed tons of private libavformat symbols all over the place. Don't do this and convert all code to proper public APIs. As a consequence, the code becomes shorter and cleaner (many things the code tried are done by libavformat APIs).
* audio: replace af_fmt2str_short -> af_fmt_to_strwm42013-11-071-1/+1
| | | | Also, remove all af_fmt2str usages.
* configure: uniform the defines to #define HAVE_xxx (0|1)Stefano Pigozzi2013-11-031-1/+1
| | | | | | | | | | | | | | | | | | | | | The configure followed 5 different convetions of defines because the next guy always wanted to introduce a new better way to uniform it[1]. For an hypothetic feature 'hurr' you could have had: * #define HAVE_HURR 1 / #undef HAVE_DURR * #define HAVE_HURR / #undef HAVE_DURR * #define CONFIG_HURR 1 / #undef CONFIG_DURR * #define HAVE_HURR 1 / #define HAVE_DURR 0 * #define CONFIG_HURR 1 / #define CONFIG_DURR 0 All is now uniform and uses: * #define HAVE_HURR 1 * #define HAVE_DURR 0 We like definining to 0 as opposed to `undef` bcause it can help spot typos and is very helpful when doing big reorganizations in the code. [1]: http://xkcd.com/927/ related
* demux: rename Windows symbolswm42013-11-021-1/+1
| | | | | | | | | | | | | | | | | | | | There are some Microsoft Windows symbols which are traditionally used by the mplayer core, because it used to be convenient (avi was the big format, using binary windows decoders made sense...). So these symbols have the exact same definition as the Windows one, and if mplayer is compiled on Windows, the symbols from windows.h are used. This broke recently just because some files were shuffled around, and the symbols defined in ms_hdr.h collided with windows.h ones. Since we don't have windows binary decoders anymore, there's not the slightest reason our symbols should have the same names. Rename them to reduce the risk for collision, and to fix the recent regression. Drop WAVEFORMATEXTENSIBLE, because it's mostly unused. ao_dsound defines its own version if the windows headers don't define it, and ao_wasapi is not available on systems where this symbol is missing. Also reindent ms_hdr.h.
* ad_mpeg123: support in-stream format changesThomas Orgis2013-10-061-61/+88
| | | | | | | | | git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@36461 b3059339-0415-0410-9bf9-f77b7e298cf2 Fixes playback of http://mpg123.org/test/44and22.mp3 Cherry-picked from MPlayer SVN rev. #36461, a patch by Thomas Orgis, committed by by Reimar Döffinger.
* cosmetics: replace "CTRL" defines by enumswm42013-10-021-1/+3
| | | | Because why not.
* audio: fix playback of Musepack SV8 fileswm42013-09-011-2/+4
| | | | | | | | | | | | | | This is basically a libavcodec API oddity: it can happen that avcodec_decode_audio4() returns 0 (meaning 0 bytes were consumed). It requires you to feed the complete packet again to decode the full packet, and to successfully decode the following packets. We ignored this case with the argument that there's the danger of an endless decode loop (because nothing of that packet is apparently decoded, so it would retry forever), but change it in order to decode mpc8 files correctly. Also add some comments to explain the mess.
* core: move contents to mpvcore (2/2)Stefano Pigozzi2013-08-065-13/+13
| | | | Followup commit. Fixes all the files references.
* audio/filter: use new option APIwm42013-07-222-14/+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 crapwm42013-07-227-115/+61
| | | | | Declare decoders directly, instead of using the LIBAD_EXTERN macro. This is simpler (no weird magic) and more extensible.
* demux_mkv: never force output sample ratewm42013-07-161-12/+5
| | | | | | | | | | | | | | Matroska has an output sample rate (OutputSamplingFrequency), which in theory should be forced instead of whatever the decoder outputs. But it appears no software (other than mplayer2 and mpv until now) actually respects this. Even worse, there were broken files around, which played correctly with (in theory) broken software, but not mplayer2/mpv. Hacks were added to our code to play these files correctly, but they didn't catch all cases. Simplify this by doing what everyone else does, and always use the decoder's sample rate instead. In particular, we try to handle all sample rate issues like libavformat's Matroska demuxer does.
* ad_lavc: re-unsimplify, fix libavcodec API usagewm42013-07-111-2/+19
| | | | | | | | | | | | | | | | | | | | | | It turns out that some code that was removed earlier was still needed. avcodec_decode_audio4() can decode packets "partially". In that case, you have to "slice" the packet and call the decode function again. Codecs which need this are obscure and in low numbers. One sample that needs it is here: rsync://fate-suite.ffmpeg.org/fate-suite/lossless-audio/luckynight-partial.shn (This one decodes in rather small increments.) The new code is much simpler than what has been removed earlier, though. The fact that we own the packet returned by the demuxer helps a lot. Not sure what should happen if avcodec_decode_audio4() returns 0. Currently, we throw away the packet in this case. We don't want to be stuck in an endless loop (could happen if the decoder produces no output either).
* 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.
* ad_spdif: better PTS syncwm42013-07-111-1/+3
| | | | | | | | | | | pts_bytes can't just be changed at the end. It must be offset to the pts value, which is reset with each packet read from the demuxer. Make sure the pts_byte field is always reset after receiving a new PTS, i.e. increment it after actually writing to the output buffer. Flush the AVFormatContext's write buffer, because otherwise the audio PTS will jump around too much: the calculation doesn't use the exact output buffer size if there's still data in the avio buffer.
* demux: remove facility for partial packet readswm42013-07-115-73/+23
| | | | | | | | | | | | | | | | | | 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.
* ad_lavc: simplifywm42013-07-101-30/+5
| | | | | | | | | We don't need to deal with partial packet reads, manually using an audio parser, or having to call the libavcodec decoder multiple times per packet. Actually, I'm not sure about the last point. ffplay still does this, but the ffmpeg demuxing.c example doesn't.
* audio: remove decoder input bufferwm42013-07-102-12/+0
| | | | This was unused.
* demux: remove audio parserwm42013-07-082-26/+5
| | | | | | | | | | | The audio parser was needed only by the "old" demuxers, and demux_rawaudio. All other demuxers output already parsed packets. demux_rawaudio is usually for raw audio, so using a parser with it doesn't usually make sense. But you can also force it to read compressed formats with fixed packet sizes, in which case the parser would have been used. This use case is probably broken now, but you will be able to do the same thing with libavformat demuxers.
* 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 --stereowm42013-06-132-14/+1
| | | | | | | 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.
* core: make options.c compile standalonewm42013-06-081-0/+2
| | | | | This also removes the split between "mplayer" and "common" opts (common opts used to be shared between mencoder and mplayer).
* core: add common function to initialize AVPacketwm42013-06-031-5/+2
| | | | | | | | | | Audio and video had their own (very similar) functions to initialize an AVPacket (ffmpeg's packet struct) from a demux_packet (mplayer's packet struct). Add a common function for these. Also use this function for sd_lavc_conv. This is actually a functional change, as some libavfilter subtitle demuxers add weird out-of-band stuff as side-data.
* Merge branch 'audio_changes'wm42013-05-126-88/+98
|\ | | | | | | | | Conflicts: audio/out/ao_lavc.c
| * audio: print channel map additionally to channel count on terminalwm42013-05-121-4/+5
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| * ad_lavc: force channel layout pass-through with demux_rawaudiowm42013-05-121-2/+11
| | | | | | | | | | | | | | | | Using demux_rawaudio and the --rawaudio-channels option is useful for testing channel map stuff. The libavcodec PCM decoder normalizes the channel map to ffmpeg order, though. Prevent this by forcing the original channel map when using the mp-pcm pseudo decoder entry (used by demux_rawaudio and stream/tv.c only).
| * core: use channel map on demuxer level toowm42013-05-124-21/+34
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
| * reorder_ch: remove old channel reorder functionswm42013-05-121-7/+0
| | | | | | | | | | | | | | This is done in af_lavrresample now, and as part of format negotiation. Also remove the remaining reorder_channel calls. They were redundant and did nothing.
| * audio/out: switch to channel mapwm42013-05-122-4/+7
| | | | | | | | | | | | 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.
| * options: add option to prevent decoder audio downmixingwm42013-04-131-3/+24
| | | | | | | | Also rename --a52drc to --ad-lavc-ac3drc, and add --ad-lavc-o.
| * 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 optionwm42013-04-132-16/+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.
| * ad_lavc: use fmt-conversion to map sample formatsStefano Pigozzi2013-04-131-18/+6
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* | audio/decode: remove vararg from ad_control()wm42013-04-125-5/+5
|/ | | | | This was unused and dumb. Ancient MPlayer used varargs instead of void* arguments for control() functions, and this was the last leftover.
* Prefix CODEC_ID_ with AV_wm42013-03-132-15/+15
| | | | | | | | | The old names have been deprecated a while ago, but were needed for supporting older ffmpeg/libav versions. The deprecated identifiers have been removed from recent Libav and FFmpeg git. This change breaks compatibility with Libav 0.8.x and equivalent FFmpeg releases.
* Add a --dtshd optionWessel Dankers2013-03-041-13/+22
| | | | | | | | | | The spdif decoder was hardcoded to assume that the spdif output is capable of accepting high (>1.5Mbps) bitrates. While this is true for modern HDMI spdif interfaces, the original coax/toslink system cannot deal with this and will fail to work. This patch adds an option --dtshd which can be enabled if you use a DTS-capable receiver behind a HDMI link.
* demux_lavf, ad_lavc, vd_lavc: pass codec header data directlywm42013-02-101-8/+10
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