aboutsummaryrefslogtreecommitdiffstats
Commit message (Collapse)AuthorAgeFilesLines
...
* configure: move ranlib -D test after setting defaultsAdrian Ratiu2022-03-111-5/+5
| | | | | | | | | | | | | | | | | In Gentoo and ChromeOS we want to allow pure LLVM builds without using GNU tools, so we block any unwanted mixed GNU/LLVM usages (GNU tools are still kept around in our chroots for projects like glibc which cannot yet be built otherwise). The default ${cross_prefix}${ranlib_default} points to GNU and fails, so move the test a bit later - after the defaults are set and the proper values get overriden - such that ffmpeg configure calls the llvm-ranlib we desire. [1] [1] https://gitweb.gentoo.org/repo/gentoo.git/tree/media-video/ffmpeg/ffmpeg-4.4.1-r1.ebuild?id=7a34377e3277a6a0e2eedd40e90452a44c55f1e6#n477 Signed-off-by: Adrian Ratiu <adrian.ratiu@collabora.com> Signed-off-by: Martin Storsjö <martin@martin.st>
* avfilter/vf_zscale: do not attempt to continue filtering if there is no graphPaul B Mahol2022-03-111-5/+22
|
* avfilter/vf_zscale: also check formatsPaul B Mahol2022-03-111-1/+2
|
* avutil/cpu: add AVX512 Icelake flagWu Jianhua2022-03-109-50/+67
| | | | | | Signed-off-by: Wu Jianhua <jianhua.wu@intel.com> Reviewed-by: Henrik Gramner <henrik@gramner.com> Signed-off-by: James Almer <jamrial@gmail.com>
* avformat: add DFPWM WAV container supportJack Bruienne2022-03-103-3/+6
| | | | | | | | | | | | | | | | | | | This commit adds support for storing DFPWM audio in a WAV container. It uses the WAVEFORMATEXTENSIBLE structure, following these conventions: https://gist.github.com/MCJack123/90c24b64c8e626c7f130b57e9800962c The implementation is very simple: it just adds the GUID to the list of WAV GUIDs, and modifies the WAV muxer to always use WAVEFORMATEXTENSIBLE format with that GUID. This creates a standard container format for DFPWM besides raw data. It will allow users to transfer DFPWM audio in a standard container format, with the sample rate and channel count contained in the file as opposed to being an external parameter as in the raw format. This format is already supported in my AUKit library, which is the CC analog to libav (albeit much smaller). Support in other applications is TBD. Signed-off-by: Jack Bruienne <jackbruienne@gmail.com>
* avformat: add DFPWM raw formatJack Bruienne2022-03-108-3/+104
| | | | | | | | | | | | | This patch builds on my previous DFPWM codec patch, adding a raw audio format to be able to read/write the raw files that are most commonly used (as no other container format supports it yet). The muxers are mostly copied from the PCM demuxer and the raw muxers, as DFPWM is typically stored as raw data. Please see the previous patch for more information on DFPWM. Signed-off-by: Jack Bruienne <jackbruienne@gmail.com>
* avcodec: add DFPWM1a codecJack Bruienne2022-03-1011-1/+273
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | From the wiki page (https://wiki.vexatos.com/dfpwm): > DFPWM (Dynamic Filter Pulse Width Modulation) is an audio codec > created by Ben “GreaseMonkey” Russell in 2012, originally to be used > as a voice codec for asiekierka's pixmess, a C remake of 64pixels. > It is a 1-bit-per-sample codec which uses a dynamic-strength one-pole > low-pass filter as a predictor. Due to the fact that a raw DPFWM decoding > creates a high-pitched whine, it is often followed by some post-processing > filters to make the stream more listenable. It has recently gained popularity through the ComputerCraft mod for Minecraft, which added support for audio through this codec, as well as the Computronics expansion which preceeded the official support. These both implement the slightly adjusted 1a version of the codec, which is the version I have chosen for this patch. This patch adds a new codec (with encoding and decoding) for DFPWM1a. The codec sources are pretty simple: they use the reference codec with a basic wrapper to connect it to the FFmpeg AVCodec system. To clarify, the codec does not have a specific sample rate - it is provided by the container (or user), which is typically 48000, but has also been known to be 32768. The codec does not specify channel info either, and it's pretty much always used with one mono channel. However, since it appears that libavcodec expects both sample rate and channel count to be handled by either the codec or container, I have made the decision to allow multiple channels interleaved, which as far as I know has never been used, but it works fine here nevertheless. The accompanying raw format has a channels option to set this. (I expect most users of this will not use multiple channels, but it remains an option just in case.) This patch will be highly useful to ComputerCraft developers who are working with audio, as it is the standard format for audio, and there are few user-friendly encoders out there, and even fewer decoders. It will streamline the process for importing and listening to audio, replacing the need to write code or use tools that require very specific input formats. You may use the CraftOS-PC program (https://www.craftos-pc.cc) to test out DFPWM playback. To use it, run the program and type this command: "attach left speaker" Then run "speaker play <file.dfpwm>" for each file. The app runs in a sandbox, so files have to be transferred in first; the easiest way to do this is to simply drag the file on the window. (Or copy files to the folder at https://www.craftos-pc.cc/docs/saves.) Sample DFPWM files can be generated with an online tool at https://music.madefor.cc. This is the current best way to encode DFPWM files. Simply drag an audio file onto the page, and it will encode it, giving a download link on the page. I've made sure to update all of the docs as per Developer§7, and I've tested it as per section 8. Test files encoded to DFPWM play correctly in ComputerCraft, and other files that work in CC are correctly decoded. I have also verified that corrupt files do not crash the decoder - this should theoretically not be an issue as the result size is constant with respect to the input size. Signed-off-by: Jack Bruienne <jackbruienne@gmail.com>
* avfilter/af_lv2: add commands supportPaul B Mahol2022-03-102-0/+25
|
* avfilter/af_anlmdn: add support for using writable framesPaul B Mahol2022-03-101-13/+23
|
* fate: add a test for HDR Vivid metadata in HEVCLimin Wang2022-03-102-0/+81
| | | | Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
* avfilter/af_anlmdn: stop using fifo and rewriting ptsPaul B Mahol2022-03-091-101/+62
|
* avfilter/af_anlmdn: fix possible array overflow and increase options limitsPaul B Mahol2022-03-092-19/+24
|
* arm64: Add Armv8.3-A PAC support to assembly filesAndre Kempe2022-03-094-11/+113
| | | | | | | | | | | This patch adds optional support for Arm Pointer Authentication Codes. PAC support is turned on or off at compile time using additional compiler flags. Unless any of these is enabled explicitly, no additional code will be emitted at all. Signed-off-by: André Kempe <andre.kempe@arm.com> Signed-off-by: Martin Storsjö <martin@martin.st>
* lavfi/drawtext: Add %N for drawing fractions of a secondThilo Borgmann2022-03-082-3/+71
| | | | Suggested-By: ffmpeg@fb.com
* avfilter/f_segment: fix sending frames with zero samples outPaul B Mahol2022-03-081-2/+15
| | | | Fix max_samples variable type, and check for out of range values.
* avcodec/pngdec: support alpha blending for palette apngPaul B Mahol2022-03-082-69/+72
| | | | Update clock test, as PAL8 apngs are now decoded as RGBA.
* avcodec/argo: Check packet sizeMichael Niedermayer2022-03-081-0/+3
| | | | | | | | Fixes: Timeout Fixes: 45052/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ARGO_fuzzer-6033489206575104 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avcodec/g729_parser: Check channelsMichael Niedermayer2022-03-081-0/+3
| | | | | | | | | Fixes: signed integer overflow: 10 * 808464428 cannot be represented in type 'int' Fixes: assertion failure Fixes: ticket9651 Reviewed-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avformat/avidec: Check heightMichael Niedermayer2022-03-081-0/+2
| | | | | | | Fixes: negation of -2147483648 cannot be represented in type 'int'; cast to an unsigned type to negate this value to itself Fixes: Ticket8486 Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* tools/target_dec_fuzzer: Adjust threshold for targaMichael Niedermayer2022-03-081-0/+1
| | | | | | | | Fixes: Timeout Fixes: 44877/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_TARGA_fuzzer-4870505251864576 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avformat/rmdec: Better duplicate tags checkMichael Niedermayer2022-03-081-4/+5
| | | | | | | | Fixes: memleaks Fixes: 44810/clusterfuzz-testcase-minimized-ffmpeg_dem_IVR_fuzzer-5619494647627776 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avfilter/f_sendcmd: export width and height tooPaul B Mahol2022-03-072-0/+12
|
* avfilter/af_channelsplit: switch to activate()Paul B Mahol2022-03-071-18/+61
|
* avfilter/af_acrossover: switch to activate()Paul B Mahol2022-03-071-12/+56
|
* avfilter/vf_extractplanes: switch to activate()Paul B Mahol2022-03-071-37/+68
| | | | | | | Fixes hang at end of input with this command: ffmpeg -f lavfi -i testsrc2=d=50,format=yuv444p -lavfi \ "extractplanes=y+u+v[y][u][v];[y]tpad=start=0[y];[u]tpad=start=0[u];[v]negate[v];[y][u][v]vstack=3" -f null -
* avfilter/split: switch to activate()Paul B Mahol2022-03-071-17/+51
|
* libavfilter: vf_scale: Properly take in->color_range into accountMartin Storsjö2022-03-071-0/+13
| | | | | | | | | | | | | | | | | | | | | While swscale can be reconfigured with sws_setColorspaceDetails, the in/out ranges also need to be set before calling sws_init_context, otherwise the initialization might choose fastpaths that don't take the ranges into account. Therefore, look at in->color_range too, when deciding on whether the scaler needs to be reconfigured. Add a new member variable for keeping track of this, for being able to differentiate between whether the scale filter parameter "in_range" has been set (which should override whatever the input frame has set) or whether it has been configured based on the latest frame (which should trigger reconfiguring the scaler if the input frame ranges change). Fixes: Ticket #9576 Signed-off-by: Martin Storsjö <martin@martin.st>
* avfilter/vf_colorlevels: Fix build failure on ARMMichael Niedermayer2022-03-061-8/+8
| | | | | | | | | | This fixes building for arm after 10c2ef1ca41dbe7811f0588f4163c8cf7b8fda66. The argument to av_clip_uintp2 must be an assembly time immediate constant. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by and commit message details-by: Martin Storsjö <martin@martin.st> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avfilter/vsrc_gradients: add radial gradientsPaul B Mahol2022-03-062-9/+16
|
* avfilter/vsrc_gradients: add gbrapf32 format supportPaul B Mahol2022-03-061-3/+94
|
* avfilter/vf_colorlevels: add planar rgb formats supportPaul B Mahol2022-03-061-43/+189
|
* avcodec: add pcm-bluray encoderPaul B Mahol2022-03-065-2/+302
|
* avformat/mpegtsenc: fix muxing pcm-blurayPaul B Mahol2022-03-062-1/+5
|
* tests: update hash as output have changed again for fate-lavf-mxf_opatomPaul B Mahol2022-03-061-1/+1
|
* avfilter/avf_ahistogram: use av_clip_uint8() insteadPaul B Mahol2022-03-061-4/+4
|
* avcodec/dnxhdenc: fill unused bytes from put bits buffer with zerosPaul B Mahol2022-03-051-0/+1
|
* avfilter/avf_ahistogram: add new histogram mode optionPaul B Mahol2022-03-052-10/+69
|
* avcodec/dnxhdenc: retry increasing qscale to not overflow max_bitsPaul B Mahol2022-03-053-6/+8
| | | | | | | Increase mb_bits type from uint16_t to uint32_t to fix possible overflows in bit size calculations. Update fate test that needs change.
* avcodec/dnxhdenc: fix possible out of bound writes for big w/hPaul B Mahol2022-03-042-1/+4
| | | | It was caused by integer overflows.
* fftools/ffmpeg_opt: Apply copyinkf for all stream typesAndreas Rheinhardt2022-03-041-4/+3
| | | | | | | | | | The earlier code has ignored it for all stream types except video and subtitles, probably because audio was presumed to only consist of keyframes. Yet this assumption is not true for e.g. TrueHD. Reviewed-by: Jan Ekström <jeebjp@gmail.com> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
* avfilter/vf_pad: use already available outlink variablePaul B Mahol2022-03-041-2/+2
|
* avformat/mov: reindent after previous commitClément Bœsch2022-03-041-6/+6
|
* avformat/mov: fix seeking with HEVC open GOP filesClément Bœsch2022-03-042-0/+125
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This was tested with medias recorded from an iPhone XR and an iPhone 13. Here is how a typical stream looks like in coding order: ┌────────┬─────┬─────┬──────────┐ │ sample | PTS | DTS | keyframe | ├────────┼─────┼─────┼──────────┤ ┊ ┊ ┊ ┊ ┊ │ 53 │ 560 │ 510 │ No │ │ 54 │ 540 │ 520 │ No │ │ 55 │ 530 │ 530 │ No │ │ 56 │ 550 │ 540 │ No │ │ 57 │ 600 │ 550 │ Yes │ │ * 58 │ 580 │ 560 │ No │ │ * 59 │ 570 │ 570 │ No │ │ * 60 │ 590 │ 580 │ No │ │ 61 │ 640 │ 590 │ No │ │ 62 │ 620 │ 600 │ No │ ┊ ┊ ┊ ┊ ┊ In composition/display order: ┌────────┬─────┬─────┬──────────┐ │ sample | PTS | DTS | keyframe | ├────────┼─────┼─────┼──────────┤ ┊ ┊ ┊ ┊ ┊ │ 55 │ 530 │ 530 │ No │ │ 54 │ 540 │ 520 │ No │ │ 56 │ 550 │ 540 │ No │ │ 53 │ 560 │ 510 │ No │ │ * 59 │ 570 │ 570 │ No │ │ * 58 │ 580 │ 560 │ No │ │ * 60 │ 590 │ 580 │ No │ │ 57 │ 600 │ 550 │ Yes │ │ 63 │ 610 │ 610 │ No │ │ 62 │ 620 │ 600 │ No │ ┊ ┊ ┊ ┊ ┊ Sample/frame 58, 59 and 60 are B-frames which actually depends on the key frame (57). Here the key frame is not an IDR but a "CRA" (Clean Random Access). Initially, I thought I could rely on the sdtp box (independent and disposable samples), but unfortunately: sdtp[54] is_leading:0 sample_depends_on:1 sample_is_depended_on:0 sample_has_redundancy:0 sdtp[55] is_leading:0 sample_depends_on:1 sample_is_depended_on:2 sample_has_redundancy:0 sdtp[56] is_leading:0 sample_depends_on:1 sample_is_depended_on:2 sample_has_redundancy:0 sdtp[57] is_leading:0 sample_depends_on:2 sample_is_depended_on:0 sample_has_redundancy:0 sdtp[58] is_leading:0 sample_depends_on:1 sample_is_depended_on:0 sample_has_redundancy:0 sdtp[59] is_leading:0 sample_depends_on:1 sample_is_depended_on:2 sample_has_redundancy:0 sdtp[60] is_leading:0 sample_depends_on:1 sample_is_depended_on:2 sample_has_redundancy:0 sdtp[61] is_leading:0 sample_depends_on:1 sample_is_depended_on:0 sample_has_redundancy:0 sdtp[62] is_leading:0 sample_depends_on:1 sample_is_depended_on:0 sample_has_redundancy:0 The information that might have been useful here would have been is_leading, but all the samples are set to 0 so this was unusable. Instead, we need to rely on sgpd/sbgp tables. In my case the video track contained 3 sgpd tables with the following grouping types: tscl, sync and tsas. In the sync table we have the following 2 entries (only): sgpd.sync[1]: sync nal_unit_type:0x14 sgpd.sync[2]: sync nal_unit_type:0x15 (The count starts at 1 because 0 carries the undefined semantic, we'll see that later in the reference table). The NAL unit types presented here correspond to: libavcodec/hevc.h: HEVC_NAL_IDR_N_LP = 20, libavcodec/hevc.h: HEVC_NAL_CRA_NUT = 21, In parallel, the sbgp sync table contains the following: ┌────┬───────┬─────┐ │ id │ count │ gdi │ ├────┼───────┼─────┤ │ 0 │ 1 │ 1 │ │ 1 │ 56 │ 0 │ │ 2 │ 1 │ 2 │ │ 3 │ 59 │ 0 │ │ 4 │ 1 │ 2 │ │ 5 │ 59 │ 0 │ │ 6 │ 1 │ 2 │ │ 7 │ 59 │ 0 │ │ 8 │ 1 │ 2 │ │ 9 │ 59 │ 0 │ │ 10 │ 1 │ 2 │ │ 11 │ 11 │ 0 │ └────┴───────┴─────┘ The gdi column (group description index) directly refers to the index in the sgpd.sync table. This means the first frame is an IDR, then we have batches of undefined frames interlaced with CRA frames. No IDR ever appears again (tried on a 30+ seconds sample). With that information, we can build an heuristic using the presentation order. A few things needed to be introduced in this commit: 1. min_sample_duration is extracted from the stts: we need the minimal step between sample in order to PTS-step backward to a valid point 2. In order to avoid a loop over the ctts table systematically during a seek, we build an expanded list of sample offsets which will be used to translate from DTS to PTS 3. An open_key_samples index to keep track of all the non-IDR key frames; for now it only supports HEVC CRA frames. We should probably add BLA frames as well, but I don't have any sample so I prefered to leave that for later It is entirely possible I missed something obvious in my approach, but I couldn't come up with a better solution. Also, as mentioned in the diff, we could optimize is_open_key_sample(), but the linear scaling overhead should be fine for now since it only happens in seek events. Fixing this issue prevents sending broken packets to the decoder. With FFmpeg hevc decoder the frames are skipped, with VideoToolbox the frames are glitching.
* avformat/mov: add parsing for the sgpd sync boxClément Bœsch2022-03-042-0/+61
| | | | | | | | | | | sgpd means Sample Group Description Box. For now, only the sync grouping type is parsed, but the function can easily be adjusted to support other flavours. The sbgp (Sample to Group Box) sync_group table built in previous commit contains references to this table through the group_description_index field.
* avformat/mov: add support for sync group in sbgp boxClément Bœsch2022-03-042-0/+7
|
* avformat/mov: prepare sbgp parsing for other grouping typesClément Bœsch2022-03-041-11/+19
|
* avfilter/vf_zscale: fix leaks in fast/bypass pathPaul B Mahol2022-03-041-11/+9
|
* avfilter/avf_abitscope: make frame writable before writing to itPaul B Mahol2022-03-041-0/+1
|
* avfilter/avf_aphasemeter: make frame writable before writing to itPaul B Mahol2022-03-041-0/+1
|
* avfilter/avf_ahistogram: make frame writable before writing to itPaul B Mahol2022-03-041-0/+1
|
generated by cgit v1.2.3 (git 2.25.1) at 2025-07-15 20:23:31 +0000