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* stream: split out pthread helper functionwm42013-11-174-26/+70
| | | | Also split the function itself into 3.
* af_lavrresample: set cutoff as double, not intwm42013-11-171-1/+1
| | | | Regression introduced with commit a89549e8.
* player: write final audo chunk only if there is audio leftwm42013-11-171-4/+6
| | | | | Don't call ao_play() if there's nothing left. Of course this still asks the AO to play internally buffered audio by setting drain=true.
* ao_null: properly simulate final chunk, add buffer optionswm42013-11-172-19/+56
| | | | | | | | | | Simulate proper handling of AOPLAY_FINAL_CHUNK. Print when underruns occur (i.e. running out of data). Add some options that control simulated buffer and outburst sizes. All this is useful for debugging and self-documentation. (Note that ao_null always was supposed to simulate an ideal AO, which is the reason why it fools people who try to use it for benchmarking video.)
* player: don't remove playback status when reinitializing DVBwm42013-11-171-1/+3
| | | | Also break that line a bit.
* demux_packet: add source stream indexwm42013-11-162-0/+5
| | | | Might be helpful later.
* demux: update a commentwm42013-11-161-3/+3
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* demux: remove unused commandswm42013-11-161-2/+0
| | | | These were replaced with DEMUXER_CTRL_SWITCHED_TRACKS a while ago.
* demux: simplify handling of filepos fieldwm42013-11-165-10/+13
| | | | | | | | | | | | demuxer->filepos contains the byte offset of the last read packet. This is so that the player can estimate the current playback position, if no proper timestamps are available. Simplify it to use demux_packet->pos in the generic demuxer code, instead of bothering every demuxer implementation about it. (Note that this is still a bit incorrect: it relfects the position of the last packet read by the demuxer, not that returned to the user. But that was already broken, and is not that trivial to fix.)
* demux_lavf: remove broken and commented byte based seekswm42013-11-161-40/+0
| | | | | | | | | | | | | This was originally added for better seeking where libavformat's seek function won't work well: files with timestamp resets. In these cases, the code tried to calculate an average bitrate, and then do byte based seeks by multiplying the seek target time with the bitrate. Apparently this was unreliable enough that the code was just commented (and other parts became inactive). Get rid of it. Note that the player still does byte based seeks in these cases when doing percent-seeks.
* demux: reset EOF flag differentlywm42013-11-161-10/+9
| | | | | This should be almost equivalent, but is slightly better because the EOF flag is reset earlier.
* encode_lavc: add missing newline in error messagewm42013-11-161-1/+1
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* ao_lavc: use af_format_conversion_score()wm42013-11-161-26/+29
| | | | | | | | | | This should allow it to select better fallback formats, instead of picking the first encoder sample format if ao->format is not equal to any of the encoder sample formats. Not sure what is supposed to happen if the encoder provides no compatible sample format (or no sample format list at all), but in this case ao_lavc.c still fails gracefully.
* audio/format: add heuristic to estimate loss on format conversionwm42013-11-162-0/+51
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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/format: fix doublep sample formatwm42013-11-161-1/+1
| | | | This was accidentally equivalent to floatp.
* ao_lavc: write the final audio chunks from uninit()Rudolf Polzer2013-11-161-7/+10
| | | | | | | | | These must be written even if there was no "final frame", e.g. due to the player being exited with "q". Although the issue is mostly of theoretical nature, as most audio codecs don't need the final encoding calls with NULL data. Maybe will be more relevant in the future.
* ao_lavc: fix crash with interleaved audio outputs.Rudolf Polzer2013-11-161-2/+4
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* audio: drop "_NE"/"ne" suffix from audio formatswm42013-11-1529-75/+63
| | | | | | 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.
* manpage: mark DTS-HD passthough as brokenwm42013-11-151-0/+2
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* dec_audio: adjust "large" decoding amountwm42013-11-151-5/+5
| | | | | | | | | | This used to be in bytes, now it's in samples. Divide the value by 8 (assuming a typical audio format, float samples with 2 channels). Fix some editing mistake or non-sense about the extra buffering added (1<<x instead of x<<5). Also sneak in a s/MPlayer/mpv/.
* mp_ring: remove unused functionwm42013-11-152-47/+0
| | | | This was needed for ao_jack.c., but not anymore.
* af_lavcac3enc: use option parserwm42013-11-152-43/+46
| | | | | | | This changes option parsing as well as filter defaults slightly. The default is now to encode to spdif (this is way more useful than writing raw AC3 - what was this even useful for, other than writing broken ac3 -in-wav files?). The bitrate parameter is now always in kbps.
* ad_spdif: fix regressionswm42013-11-142-9/+9
| | | | | | | | | | Apparently this was completely broken after commit 22b3f522. Basically, this locked up immediately completely while decoding the first packet. The reason was that the buffer calculations confused bytes and number of samples. Also, EOF reporting was broken (wrong return code). The special-casing of ad_mpg123 and ad_spdif (with DECODE_MAX_UNIT) is a bit annoying, but will eventually be solved in a better way.
* osx bundle: remove embedded fonts.confStefano Pigozzi2013-11-141-120/+0
| | | | | | This could cause the bundle to recache stuff because of differences with configuration of other software using fonconfig. The defaults OS X directories should be added to fontconfig at build time (through configure).
* ao_alsa: non-interleaved access is not always availablewm42013-11-141-0/+5
| | | | | | I thought this would always work... how disappointing. Revert to interleaved format if requesting non-interleaved fails.
* demux: use talloc for certain stream headerswm42013-11-144-49/+21
| | | | | | | Slightly simplifies memory management. This might make adding a demuxer cache wrapper easier at a later point, because you can just copy the complete stream header, without worrying that the wrapper will free the individual stream header fields.
* audio: fix audio data memory leakwm42013-11-141-1/+1
| | | | | Practically all audio decoding and filtering code leaked sample data memory after uninitialization due to a simple logic bug (or typo).
* gl_common: print SW renderer warning only if it was the only reason we ↵wm42013-11-141-1/+1
| | | | rejected it
* vd_lavc: select correct hw decoder profile for constrained baseline h264wm42013-11-141-2/+4
| | | | | | | | | | | | | | | | | | | | The existing code tried to remove the "extra" profile flags for h264. FF_PROFILE_H264_INTRA doesn't matter for us at all, because it's set only for profiles the vdpau/vaapi APIs don't support. The FF_PROFILE_H264_CONSTRAINED flag on the other hand is added to H264_BASELINE, except that it makes the file a real subset of H264_MAIN and H264_HIGH. Removing that flag would select the BASELINE profile, which appears to be rarely supported by hardware decoders. This means we accidentally rejected perfectly hardware decodable files. Use MAIN for it instead. (vaapi has explicit support for CONSTRAINED_BASELINE, but it seems to be a new thing, and is not reported as supported where I tried. So don't bother to check it, and do the same as on vdpau.) See github issue #204.
* gl_common: remove unneeded callbackwm42013-11-144-4/+0
| | | | We got rid of this some time ago, but apparently not completely.
* tvi_v4l2: remove VBI stuffwm42013-11-131-100/+0
| | | | | | | | This used to be needed for teletext support. Teletext commit has been removed (see commit ebaaa41f), and it appears this code is inactive. It was just forgotten with the removal. Get rid of it completely. Untested. (Like all changes to the TV code.)
* configure: enable v4l2 input on freebsdbugmen0t2013-11-131-2/+4
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* tvi_v4l2: let libv4l2 convert to a known pixel formatbugmen0t2013-11-132-47/+77
| | | | | | | | | | | Signed-off-by: wm4 <wm4@nowhere> Significant modifications over the original patch by not overriding syscalls with macros ("#define open v4l2open") for fallback, but the other way around ("#define v4l2open open"). As consequence, the calls have to be replaced throughout the file. Untested, although the original patch probably was tested.
* stream: don't include linux/types.h in some fileswm42013-11-133-4/+0
| | | | | | Apparently this is not portable to FreeBSD. It turns out that we (probably) don't use any symbols defined by this header directly, so the includes are not needed.
* m_option: handle audio/filter filters with old option parsingwm42013-11-131-3/+9
| | | | | | | | | These use the _oldargs_ hack, which failed in combination with playback resume. Make it work. It would be better to port all filters to new option parsing, but that's obviously too much work, and most filters will probably be deleted and replaced by libavfilter in the long run.
* ao_null: add untimed sub-optionwm42013-11-132-3/+24
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* ao_null: support pausing properlywm42013-11-131-4/+14
| | | | | | ao_null should simulate a "perfect" AO, but framestepping behaved quite badly with it. Framstepping usually exposes problems with AOs dropping their buffers on pause, and that's what happened here.
* mf: silence compilation warningwm42013-11-132-3/+3
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* ao_lavc: support non-interleaved audiowm42013-11-133-232/+42
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* wayland: create xkbcommon keymap from stringAlexander Preisinger2013-11-131-2/+1
| | | | Fixes a problem where the passed size doesn't match the actuall string.
* Merge branch 'planar_audio'wm42013-11-1265-1489/+1549
|\ | | | | | | | | Conflicts: audio/out/ao_lavc.c
| * audio: add support for using non-interleaved audio from decoders directlywm42013-11-1210-495/+324
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 ifdefferywm42013-11-122-50/+3
| | | | | | | | 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.
| * mp_audio: use av_malloc (cargo cult for libav*)wm42013-11-122-3/+30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | libav* is generally freaking horrible, and might do bad things if the data pointer passed to it are not aligned. One way to be sure that the alignment is correct is allocating all pointers using av_malloc(). It's possible that this is not needed at all, though. For now it might be better to keep this, since the mp_audio code is intended to replace another buffer in dec_audio.c, which is currently av_malloc() allocated. The original reason why this uses av_malloc() is apparently because libavcodec used to directly encode into mplayer buffers, which is not the case anymore, and thus (probably) doesn't make sense anymore. (The commit subject uses the word "cargo cult", after all.)
| * ao_jack: switch from interleaved to planar audioWilliam Light2013-11-121-95/+92
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| * ao_jack: refactoring, also fix "no-connect" optionWilliam Light2013-11-121-57/+97
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| * af_lavcac3enc: use planar formatswm42013-11-121-134/+82
| | | | | | | | | | | | | | | | | | | | | | Remove the awkward planarization. It had to be done because the AC3 encoder requires planar formats, but now we support them natively. Try to simplify buffer management with mp_audio_buffer. Improve checking for buffer overflows and out of bound writes. In theory, these shouldn't happen due to AC3 fixed frame sizes, but being paranoid is better.
| * af_lavcac3enc: simplify format negotiationwm42013-11-121-28/+33
| | | | | | | | | | | | | | | | | | | | | | | | | | The format negotiation is the same, except don't confusingly copy the input format into af->data, just to overwrite it later. af->data should alwass contain the output format, and the existing code was just a very misguided use of the af_test_output() helper function. Just set af->data to the output format immediately, and modify the input format properly. Also, if format negotiation fails (and needs another iteration), don't initialize the libavcodec encoder.
| * audio/filter: fix mul/delay scale and valueswm42013-11-1229-56/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | 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.)
| * ao_openal: support non-interleaved outputwm42013-11-121-18/+11
| | | | | | | | | | | | Since ao_openal simulates multi-channel audio by placing a bunch of mono-sources in 3D space, non-interleaved audio is a perfect match for it. We just have to remove the interleaving code.
| * ao_alsa: support non-interleaved audiowm42013-11-121-25/+23
| | | | | | | | | | | | | | | | ALSA supports non-interleaved audio natively using a separate API function for writing audio. (Though you have to tell it about this on initialization.) ALSA doesn't have separate sample formats for this, so just pretend to negotiate the interleaved format, and assume that all non-interleaved formats have an interleaved companion format.
| * ao_null: support non-interleaved audiowm42013-11-121-17/+20
| | | | | | | | Simply change internals from using byte counts to sample counts.
| * audio: switch output to mp_audio_bufferwm42013-11-126-113/+112
| | | | | | | | | | | | 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: add mp_audio_bufferwm42013-11-123-0/+197
| | | | | | | | | | | | Implementation wise, this could be much improved, such as using a ringbuffer that doesn't require copying data all the time. This is why we don't use mp_audio directly instead of mp_audio_buffer.
| * audio/out: prepare for non-interleaved audiowm42013-11-1217-85/+92
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This comes with two internal AO API changes: 1. ao_driver.play now can take non-interleaved audio. For this purpose, the data pointer is changed to void **data, where data[0] corresponds to the pointer in the old API. Also, the len argument as well as the return value are now in samples, not bytes. "Sample" in this context means the unit of the smallest possible audio frame, i.e. sample_size * channels. 2. ao_driver.get_space now returns samples instead of bytes. (Similar to the play function.) Change all AOs to use the new API. The AO API as exposed to the rest of the player still uses the old API. It's emulated in ao.c. This is purely to split the commits changing all AOs and the commits adding actual support for outputting N-I audio.
| * af: don't require filters to allocate af_instance->data, redo bufferswm42013-11-1227-211/+36
| | | | | | | | | | | | | | | | | | | | | | | | | | Allocate af_instance->data in generic code before filter initialization. Every filter needs af->data (since it contains the output configuration), so there's no reason why every filter should allocate and free it. Remove RESIZE_LOCAL_BUFFER(), and replace it with mp_audio_realloc_min(). Interestingly, most code becomes simpler, because the new function takes the size in samples, and not in bytes. There are larger change in af_scaletempo.c and af_lavcac3enc.c, because these had copied and modified versions of the RESIZE_LOCAL_BUFFER macro/function.
| * af_lavfi: add support for non-interleaved audiowm42013-1