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* audio: add af_select_best_samplerate functionKevin Mitchell2016-03-171-0/+31
| | | | | | This function chooses the best match to a given samplerate from a provided list. This can be used, for example, by the ao to decide what samplerate to use for output.
* audio: fix af_fmt_change_bytes() with spdif formatswm42015-11-071-1/+1
| | | | | | | | | This could accidentally change some spdif formats to AAC (because AAC is the first on the list and will match first). spdif formats are inherently uninterchangeable, so treat them as their own class of formats (like int vs. float). Might fix some issues with ao_wasapi.c.
* audio/format: revise af_format_conversion_scoreKevin Mitchell2015-09-101-8/+13
| | | | | | | | | | | | | | | | | | | * (de)planarize -1 * pad 1 byte -8 * truncate 1 byte -1024 * float -> int 1048576 * (8 - dst_bytes) * int -> float -512 Now the score is negative if and only if the conversion is lossy (e.g. previously s24 -> float was given a negative (lossy) score), However, int->float is still considered bad (s16->float is worse than than s16->s32). This penalizes any loss of precision more than performance / bandwidth hits. For example, previously s24->s16p was considered equal to s24->u8. Finally, we penalize padding more than (de)planarizing as this will increase the output size for example with ao_lavc.
* ao_lavc: use new sample format determination codewm42015-09-101-3/+1
| | | | | | | This is just a refactor, which makes it use the previously introduced function, and allows us to make af_format_conversion_score() private. (We drop 2 unlikely warning messages too... who cares.)
* audio/format: add function for determining sample conversion candidateswm42015-09-101-0/+31
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* audio/format: fix interlaved vs. non-interleaved conversionswm42015-09-101-1/+1
| | | | | | | | | | This mixed up the returned score for some interleaved/non-interleaved comparisons. Changing interleaving subtracted 1 point, while extending sample size by 1 byte also subtracted 1 point. (This scoring system is not ideal - it'd be much cleaner to do a 3-way sample format comparison instead, and sort the formats according to the comparison instead of the score.)
* audio/format: actually prefer float over double sample formatwm42015-09-101-1/+1
| | | | | ...for int->float conversions. This code accidentally inverted the condition.
* audio: fix format function consistency issueswm42015-06-261-23/+24
| | | | | | | | | | | Replace all the check macros with function calls. Give them all the same case and naming schema. Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes(). Introduce af_fmt_is_pcm(), and use it in situations that used !AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format was. It simply meant "not PCM".
* audio: replace format name tablewm42015-06-261-38/+19
| | | | Having a big switch() is simpler.
* audio: remove bitmask format definition messwm42015-06-261-37/+51
| | | | | | | | | | Audio formats used a semi-clever schema to encode the properties of the PCM encoding as bitfields into the format integer value. The af_fmt_change_bits() implementation becomes a bit weird, but it's an improvement to the rest of the code. (I've always disliked it, so why not get rid of it.)
* audio: remove S8, U16, U24, U32 formatswm42015-06-161-8/+7
| | | | | | | | | | | | | They are useless. Not only are they actually rarely in use; but libavcodec doesn't even output them, as libavcodec has no such sample formats for decoded audio. Even if it should happen that we actually still need them (e.g. if doing direct hardware output), there are better solutions. Swapping the sign is a fast and lossless operation and can be done inplace, so AO actually needing it could do this directly. If you wonder why we keep U8 instead of S8: because libavcodec does it.
* Update license headersMarcin Kurczewski2015-04-131-5/+4
| | | | Signed-off-by: wm4 <wm4@nowhere>
* audio: make all format query shortcuts macrosKevin Mitchell2015-04-031-12/+1
| | | | | af_fmt_is_float and af_fmt_is_planar were previously inconsistent with AF_FORAMT_IS_SPECIAL/AF_FORMAT_IS_IEC61937
* audio: fix spdif packet size unitwm42015-03-101-7/+9
| | | | | | | | | | | In commit 5f8b060e I blindly assumed that the packet sizes were in pseudo-samples, but they were actually in bytes. Oops. (The effect was that cutting the audio was a bit less precise than it can be.) Also remove the packet size from ad_spdif.c; it didn't actually use it, and simply takes what the spdif "muxer" returns.
* audio: fix spdif DTS packet sizewm42015-03-101-0/+1
| | | | Broken in one of the previous commits.
* ad_spdif: move frame sizes to a general functionwm42015-03-101-0/+13
| | | | | | Needed for the next commit. This commit should probably be reverted as soon as we're working with full audio frames internally, instead of "flat" FIFOs.
* ao/format: add af_fmt_is_floatKevin Mitchell2014-12-011-0/+5
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* audio: cleanup spdif format definitionswm42014-09-231-11/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Before this commit, there was AF_FORMAT_AC3 (the original spdif format, used for AC3 and DTS core), and AF_FORMAT_IEC61937 (used for AC3, DTS and DTS-HD), which was handled as some sort of superset for AF_FORMAT_AC3. There also was AF_FORMAT_MPEG2, which used IEC61937-framing, but still was handled as something "separate". Technically, all of them are pretty similar, but may use different bitrates. Since digital passthrough pretends to be PCM (just with special headers that wrap digital packets), this is easily detectable by the higher samplerate or higher number of channels, so I don't know why you'd need a separate "class" of sample formats (AF_FORMAT_AC3 vs. AF_FORMAT_IEC61937) to distinguish them. Actually, this whole thing is just a mess. Simplify this by handling all these formats the same way. AF_FORMAT_IS_IEC61937() now returns 1 for all spdif formats (even MP3). All AOs just accept all spdif formats now - whether that works or not is not really clear (seems inconsistent due to earlier attempts to make DTS-HD work). But on the other hand, enabling spdif requires manual user interaction, so it doesn't matter much if initialization fails in slightly less graceful ways if it can't work at all. At a later point, we will support passthrough with ao_pulse. It seems the PulseAudio API wants to know the codec type (or maybe not - feeding it DTS while telling it it's AC3 works), add separate formats for each codecs. While this reminds of the earlier chaos, it's stricter, and most code just uses AF_FORMAT_IS_IEC61937(). Also, modify AF_FORMAT_TYPE_MASK (renamed from AF_FORMAT_POINT_MASK) to include special formats, so that it always describes the fundamental sample format type. This also ensures valid AF formats are never 0 (this was probably broken in one of the earlier commits from today).
* audio: drop swapped-endian audio formatswm42014-09-231-33/+20
| | | | | | | | | | | | | | | | | | | | Until now, the audio chain could handle both little endian and big endian formats. This actually doesn't make much sense, since the audio API and the HW will most likely prefer native formats. Or at the very least, it should be trivial for audio drivers to do the byte swapping themselves. From now on, the audio chain contains native-endian formats only. All AOs and some filters are adjusted. af_convertsignendian.c is now wrongly named, but the filter name is adjusted. In some cases, the audio infrastructure was reused on the demuxer side, but that is relatively easy to rectify. This is a quite intrusive and radical change. It's possible that it will break some things (especially if they're obscure or not Linux), so watch out for regressions. It's probably still better to do it the bulldozer way, since slow transition and researching foreign platforms would take a lot of time and effort.
* audio: remove swapped-endian spdif formatswm42014-09-231-2/+2
| | | | | | | | | | | | | | | | | | | | | | IEC 61937 frames should always be little endian (little endian 16 bit words). I don't see any apparent need why the audio chain should handle swapped-endian formats. It could be that some audio outputs might want them (especially on big endian architectures). On the other hand, it's not clear how that works on these architectures, and it's not even known whether the current code works on big endian at all. If something should break, and it should turn out that swapped-endian spdif is needed on any platform/AO, swapping still could be done in-place within the affected AO, and there's no need for the additional complexity in the rest of the player. Note that af_lavcac3enc outputs big endian spdif frames for unknown reasons. Normally, the resulting data is just pulled through an auto- inserted conversion filter and turned into little endian. Maybe this was done as a trick so that the code didn't have to byte-swap the actual audio frame. In any case, just make it output little endian frames. All of this is untested, because I have no receiver hardware.
* af_fmt2bits: change to af_fmt2bps (bytes/sample) where appropriateMarcoen Hirschberg2014-05-281-8/+13
| | | | | | In most places where af_fmt2bits is called to get the bits/sample, the result is immediately converted to bytes/sample. Avoid this by getting bytes/sample directly by introducing af_fmt2bps.
* Split mpvcore/ into common/, misc/, bstr/wm42013-12-171-1/+1
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* audio/format: add heuristic to estimate loss on format conversionwm42013-11-161-0/+48
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The added function af_format_conversion_score() can be used to select the best sample format to convert to in order to reduce loss and extra conversion work. It calculates a "loss" score when going from one format to another, and for each conversion that needs to be done a certain score is subtracted. Thus, if you have to convert from one format to a set of other formats, you can calculate the score for each conversion, and pick the one with the highest score. Conversion between int and float is considered the worst case. One odd consequence is that when converting from s32 to u8 or float, u8 will be picked. Test program used to develop this follows: #define MAX_FMT 200 struct entry { const char *name; int score; }; static int compentry(const void *px1, const void *px2) { const struct entry *x1 = px1; const struct entry *x2 = px2; if (x1->score > x2->score) return 1; if (x1->score < x2->score) return -1; return 0; } int main(int argc, char *argv[]) { for (int n = 0; af_fmtstr_table[n].name; n++) { struct entry entry[MAX_FMT]; int entries = 0; for (int i = 0; af_fmtstr_table[i].name; i++) { assert(i < MAX_FMT); entry[entries].name = af_fmtstr_table[i].name; entry[entries].score = af_format_conversion_score(af_fmtstr_table[i].format, af_fmtstr_table[n].format); entries++; } qsort(&entry[0], entries, sizeof(entry[0]), compentry); for (int i = 0; i < entries; i++) { printf("%s -> %s: %d \n", af_fmtstr_table[n].name, entry[i].name, entry[i].score); } } }
* audio: drop "_NE"/"ne" suffix from audio formatswm42013-11-151-1/+0
| | | | | | You get the native format by not appending any suffix to the format. This change includes user-facing names, e.g. for the --format option.
* audio/format: add non-interleaved audio formatswm42013-11-121-0/+49
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* ao: add ao_play_silence, use for ao_alsa and ao_osswm42013-11-101-0/+6
| | | | | Also add a corresponding function to audio/format.c, which fills an audio block with silence.
* audio/format: convert format macros to enum, drop NE suffixwm42013-11-071-33/+22
| | | | | | | | | | Turn the sample format definitions into an enum. (The format bits are still macros.) The native endian versions of the new definitions don't have a NE suffix anymore, although there are still compatibility defines since too much code uses the NE variants. Rename the format bits for special formats to help to distinguish them from the actual definitions, e.g. AF_FORMAT_AC3 to AF_FORMAT_S_AC3.
* audio: replace af_fmt2str_short -> af_fmt_to_strwm42013-11-071-10/+1
| | | | Also, remove all af_fmt2str usages.
* audio/format: reformatwm42013-11-071-20/+18
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* audio/format: add some helper functionswm42013-10-221-0/+30
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* audio: don't allow setting unknown formats from command linewm42013-08-261-17/+1
| | | | | | | af_str2fmt_short(), which is used by the command line option parser, allowed passing a hex number. The user could set arbitrary integers as internal audio formats, even formats which don't exist or make no sense. This is not very useful, so get rid of it.
* audio: make internal audio format 0 an invalid formatwm42013-08-261-1/+1
| | | | | | | | | | | | Having to use -1 for that is generally quite annoying. Audio formats are created from bitmasks, and it can't be excluded that 0 is not a valid format. Fix this by adjusting AF_FORMAT_I so that it is never 0. Along with AF_FORMAT_F and the special formats, all valid formats are covered and guaranteed to be non-0. It's possible that this commit will cause some regressions, as the check for invalid audio formats changes a bit.
* audio: fix af_fmt_seconds_to_bytesStefano Pigozzi2013-06-161-2/+2
| | | | Was missing samplerate
* core: add a spsc ringbuffer implementationStefano Pigozzi2013-06-161-0/+10
| | | | | | | | | | | | | | | | | Currently every single AO was implementing it's own ringbuffer, many times with slightly different semantics. This is an attempt to fix the problem. I stole some good ideas from ao_portaudio's ringbuffer and went from there. The main difference is this one stores wpos and rpos which are absolute positions in an "infinite" buffer. To find the actual position for writing / reading just apply modulo size. The producer only modifies wpos while the consumer only modifies rpos. This makes it pretty easy to reason about and make the operations thread safe by using barriers (thread safety is guaranteed only in the Single-Producer/Single- Consumer case). Also adapted ao_coreaudio to use this ringbuffer.
* audio: add double sample formatwm42013-05-121-11/+5
| | | | | | To make this easier, get rid of the direct mapping of the AF_FORMAT_BITS_MASK bit field to number of bytes. This way we can throw away the unused AF_FORMAT_48BIT and don't have to add ..._56BIT.
* audio: add some setters for mp_audio, and require filters to use themwm42013-05-121-0/+2
| | | | | | | | | | | | | | | | 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.
* audio: remove support for native alaw/mulaw/adpcm outputwm42012-12-111-3/+0
| | | | | | This is considered a worthless feature. Note that alaw/mulaw/adpcm input is unaffected: such data is handed to libavcodec and "decoded" to linear PCM.
* Rename directories, move files (step 2 of 2)wm42012-11-121-1/+1
| | | | | | | | | | | | 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/+134
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.
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