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* avcodec/xbmenc: Do not add last comma into outputJose Da Silva2021-01-28
| | | | | | | | | | | | | | | | | | | | | | There is a minor bug in xbm encode which adds a trailing comma at the end of data. This isn't a big problem, but it would be nicer to be more technically true to an array of data (by not including the last comma). This bug fixes the output from something like this (having 4 values): static unsigned char image_bits[] = { 0x00, 0x11, 0x22, } to C code that looks like this instead (having 3 values): static unsigned char image_bits[] = { 0x00, 0x11, 0x22 } which is the intended results. Subject: [PATCH 1/3] avcodec/xbmenc: Do not add last comma into output array xbm outputs c arrays of data. Including a comma at the end means there is another value to be added. This bug fix changes something like this: static unsigned char image_bits[] = { 0x00, 0x11, 0x22, } to C code like this: static unsigned char image_bits[] = { 0x00, 0x11, 0x22 } Signed-off-by: Joe Da Silva <digital@joescat.com>
* tests/dnn: enable unit test denseGuo, Yejun2021-01-28
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* avformat/mxfenc: add Coding Equations and Color Primaries to local tagsMarton Balint2021-01-27
| | | | | | Fixes ticket #9079. Signed-off-by: Marton Balint <cus@passwd.hu>
* fate/hlsenc: rework the ffprobe dependency of hls-fmp4_ac3James Almer2021-01-25
| | | | | | | Add it to the existing FATE_SAMPLES_FFMPEG_FFPROBE list of ffprobe dependant tests instead. Signed-off-by: James Almer <jamrial@gmail.com>
* checkasm: add hevc_pel testsJosh Dekker2021-01-25
| | | | | Co-authored-by: Niklas Haas <git@haasn.xyz> Signed-off-by: Josh Dekker <josh@itanimul.li>
* tests/fate/fits: Add a todo for a 64bit test.Carl Eugen Hoyos2021-01-24
| | | | | The test should currently fail on big endian but passes because of the unsuitable input file.
* tests/fate/hlsenc: ffprobe is needed for hls-fmp4_ac3.Carl Eugen Hoyos2021-01-24
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* tests/dnn: fix build issue after function name changedGuo, Yejun2021-01-22
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* dnn-layer-conv2d-test.c: remove dependency of dnn_native_classGuo, Yejun2021-01-22
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* movenc: Present durations in mvhd/tkhd/mdhd as they are after editsMartin Storsjö2021-01-15
| | | | | | | | | | | | | | | | | | | | | If the edit lists remove parts of the output timeline, or add a delay to it, this should be included in the mvhd/tkhd/mdhd durations, which should correspond to the edit lists. For tracks starting with pts < 0, the edit list trims out the segment before pts=0. For tracks starting with pts > 0, a delay element is added in the edit list, delaying the start of the track data. In both cases, the practical effect is that the post-edit output is as if the track had started with pts = 0. Thus calculate the range from pts=0 to end_pts, for the purposes of mvhd/tkhd/mdhd, unless edit lists explicitly are disabled. mov_write_edts_tag needs to operate on the actual pts duration of the track samples, not the duration that already takes the edit list effect into account. Signed-off-by: Martin Storsjö <martin@martin.st>
* fft: remove 16-bit FFT and MDCT codeLynne2021-01-14
| | | | | | No longer used by anything. Unfortunately the old FFT_FLOAT/FFT_FIXED_32 is left as-is. It's simply too much work for code meant to be all removed anyway.
* ac3enc_fixed: convert to 32-bit sample formatLynne2021-01-14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The AC3 encoder used to be a separate library called "Aften", which got merged into libavcodec (literally, SVN commits and all). The merge preserved as much features from the library as possible. The code had two versions - a fixed point version and a floating point version. FFmpeg had floating point DSP code used by other codecs, the AC3 decoder including, so the floating-point DSP was simply replaced with FFmpeg's own functions. However, FFmpeg had no fixed-point audio code at that point. So the encoder brought along its own fixed-point DSP functions, including a fixed-point MDCT. The fixed-point MDCT itself is trivially just a float MDCT with a different type and each multiply being a fixed-point multiply. So over time, it got refactored, and the FFT used for all other codecs was templated. Due to design decisions at the time, the fixed-point version of the encoder operates at 16-bits of precision. Although convenient, this, even at the time, was inadequate and inefficient. The encoder is noisy, does not produce output comparable to the float encoder, and even rings at higher frequencies due to the badly approximated winow function. Enter MIPS (owned by Imagination Technologies at the time). They wanted quick fixed-point decoding on their FPUless cores. So they contributed patches to template the AC3 decoder so it had both a fixed-point and a floating-point version. They also did the same for the AAC decoder. They however, used 32-bit samples. Not 16-bits. And we did not have 32-bit fixed-point DSP functions, including an MDCT. But instead of templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed), they simply copy-pasted their own MDCT into ours, and completely ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected. This is also the status quo nowadays - 2 separate MDCTs, one which produces floating point and 16-bit fixed point versions, and one sort-of integrated which produces 32-bit MDCT. MIPS weren't all that interested in encoding, so they left the encoder as-is, and they didn't care much about the ifdeffery, mess or quality - it's not their problem. So the MDCT/FFT code has always been a thorn in anyone looking to clean up code's eye. Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients. So for the floating point version, the encoder simply runs the float MDCT, and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently a fixed-point codec. For the fixed-point version, the input is 16-bit samples, so to maximize precision the frame samples are analyzed and the highest set bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits, computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits. This patch simply changes the encoder to accept 32-bit samples, reusing the already well-optimized 32-bit MDCT code, allowing us to clean up and drop a large part of a very messy code of ours, as well as prepare for the future lavu/tx conversion. The coefficients are simply scaled down to 25 bits during windowing, skipping 2 separate scalings, as the hacks to extend precision are simply no longer necessary. There's no point in running the MDCT always at 32 bits when you're going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds properly. This also makes the encoder even slightly more accurate over the float version, as there's no coefficient conversion step necessary. SIZE SAVINGS: ARM32: HARDCODED TABLES: BASE - 10709590 DROP DSP - 10702872 - diff: -6.56KiB DROP MDCT - 10667932 - diff: -34.12KiB - both: -40.68KiB DROP FFT - 10336652 - diff: -323.52KiB - all: -364.20KiB SOFTCODED TABLES: BASE - 9685096 DROP DSP - 9678378 - diff: -6.56KiB DROP MDCT - 9643466 - diff: -34.09KiB - both: -40.65KiB DROP FFT - 9573918 - diff: -67.92KiB - all: -108.57KiB ARM64: HARDCODED TABLES: BASE - 14641112 DROP DSP - 14633806 - diff: -7.13KiB DROP MDCT - 14604812 - diff: -28.31KiB - both: -35.45KiB DROP FFT - 14286826 - diff: -310.53KiB - all: -345.98KiB SOFTCODED TABLES: BASE - 13636238 DROP DSP - 13628932 - diff: -7.13KiB DROP MDCT - 13599866 - diff: -28.38KiB - both: -35.52KiB DROP FFT - 13542080 - diff: -56.43KiB - all: -91.95KiB x86: HARDCODED TABLES: BASE - 12367336 DROP DSP - 12354698 - diff: -12.34KiB DROP MDCT - 12331024 - diff: -23.12KiB - both: -35.46KiB DROP FFT - 12029788 - diff: -294.18KiB - all: -329.64KiB SOFTCODED TABLES: BASE - 11358094 DROP DSP - 11345456 - diff: -12.34KiB DROP MDCT - 11321742 - diff: -23.16KiB - both: -35.50KiB DROP FFT - 11276946 - diff: -43.75KiB - all: -79.25KiB PERFORMANCE (10min random s32le): ARM32 - before - 39.9x - 0m15.046s ARM32 - after - 28.2x - 0m21.525s Speed: -30% ARM64 - before - 36.1x - 0m16.637s ARM64 - after - 36.0x - 0m16.727s Speed: -0.5% x86 - before - 184x - 0m3.277s x86 - after - 190x - 0m3.187s Speed: +3%
* fate: add tests for AVIDAnton Khirnov2021-01-01
| | | | Samples cut from tickets 971 and 4741
* api-seek-test: use non-obsolete decoding APIAnton Khirnov2021-01-01
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* api-band-test: use non-obsolete decoding APIAnton Khirnov2021-01-01
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* api-h264-test: use non-obsolete decoding APIAnton Khirnov2021-01-01
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* lavfi/vf_pp7: convert to the video_enc_params APIAnton Khirnov2021-01-01
| | | | Re-enable fate-filter-pp7
* lavfi/vf_spp: convert to the video_enc_params APIAnton Khirnov2021-01-01
| | | | Re-enable fate-filter-spp
* lavfi/vf_pp: convert to the video_enc_params APIAnton Khirnov2021-01-01
| | | | Re-enable fate-filter-qp and fate-filter-pp.
* lavfi/vf_qp: convert to the video_enc_params APIAnton Khirnov2021-01-01
| | | | Temporarily disable fate-filter-qp until vf_pp is converted.
* mpegvideo: use the AVVideoEncParams API for exporting QP tablesAnton Khirnov2021-01-01
| | | | | | | | | | Do it only when requested with the AV_CODEC_EXPORT_DATA_VIDEO_ENC_PARAMS flag. Drop previous code using the long-deprecated AV_FRAME_DATA_QP_TABLE* API. Temporarily disable fate-filter-pp, fate-filter-pp7, fate-filter-spp. They will be reenabled once these filters are converted in following commits.
* lavu: move LOCAL_ALIGNED from internal.h to mem_internal.hAnton Khirnov2021-01-01
| | | | That is a more appropriate place for it.
* fate/image: add missing ffprobe dependency to fate-dpx-probeJames Almer2020-12-18
| | | | | | And use the existing probeframes helper while at it. Signed-off-by: James Almer <jamrial@gmail.com>
* fate/image: update fate-dpx-probe reference fileJames Almer2020-12-18
| | | | | | Regression since 20b09b20a942d4aad38f9fa1324b713168d3db9a Signed-off-by: James Almer <jamrial@gmail.com>
* avcodec/dpx: Read color information from DPX headerHarry Mallon2020-12-17
| | | | Signed-off-by: Harry Mallon <harry.mallon@codex.online>
* avcodec/dpx: Report color_range from DPX headerHarry Mallon2020-12-17
| | | | Signed-off-by: Harry Mallon <harry.mallon@codex.online>
* avcodec/dpx: Read SMPTE timecode from DPXHarry Mallon2020-12-17
| | | | Signed-off-by: Harry Mallon <harry.mallon@codex.online>
* fate: Add dpx-probe testHarry Mallon2020-12-17
| | | | Signed-off-by: Harry Mallon <harry.mallon@codex.online>
* avcodec/decode: set best_effort_timestamp on output frames for all decodersJames Almer2020-12-13
| | | | | | | | | Fixes a decoding regression introduced by e9a2a87773, and as a side effect also fixes bogus values set to certain audio frames that had some samples discarded, where the offsets added to pts, pkt_dts and pkt_duration were not reflected in best_effort_timestamp. Signed-off-by: James Almer <jamrial@gmail.com>
* fate: fix fate-filter-hqx on big-endian archesAndriy Gelman2020-12-12
| | | | | | | | | | One of the inputs to the fate test has an rgba pixel format which needs to be converted to rgb32 (argb on big-endian) for the hqx filter. Because auto scaling in the fate test is disabled, this needs a separate scale filter. Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
* tests/audiomatch: add free to make static analysis tools happyJun Zhao2020-12-10
| | | | | Reviewed-by: Anton Khirnov <anton@khirnov.net> Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
* smvjpegdec: merge into mjpegdecAnton Khirnov2020-12-10
| | | | | | | | | | | | | | | | | | | | | | | SMVJPEG stores frames as slices of a big JPEG image. The decoder is implemented as a wrapper that instantiates a full internal MJPEG decoder, then forwards the decoded frames with offset data pointers. This is unnecessarily complex and fragile, not supporting useful decoder capabilities like direct rendering. Re-implement the decoder inside the MJPEG decoder, which is accomplished by returning each decoded frame multiple times, setting cropping information appropriately on each instance. One peculiar aspect of the previous design is that since - the smvjpeg decoder returns one frame per input packet - there are multiple frames in each packets (the aformentioned slices) the demuxer needs to return each packet multiple times. This is now also eliminated - the demuxer now returns each packet exactly once, with the duration set to the number of frames it decodes to. This also removes one of the last remaining internal uses of the old video decoding API.
* tests: stop using -vsync dropAnton Khirnov2020-12-10
| | | | | | | | It depends on the muxer generating the timestamps, which is deprecated and scheduled for removal on next bump. A bunch of tests change timestamps, because of ffmpeg.c is not generating them correctly. This should be fixed later.
* lavf/mux: rewrite guessing the packet durationAnton Khirnov2020-12-10
| | | | | | Factor out the code into a separate muxing-specific function. Stop accessing the deprecated AVStream-embedded codec context, use the average framerate (if specified) instead.
* tests: drop api-codec-param testAnton Khirnov2020-12-10
| | | | It fundamentally depends on deprecated lavf internals.
* avcodec/exr: preserve half-float NaN bits and add fate testMark Reid2020-12-09
| | | | | Handles NaNs more like the official implementation handles them, preserving the original bits.
* avfilter/af_earwax: fix filter behaviorPaul B Mahol2020-12-07
| | | | | Previous filter output was incorrect. New one actually follows graph in comments described on side of filter taps.
* avformat/dv: fix timestamps of audio packets in case of dropped corrupt ↵Marton Balint2020-12-06
| | | | | | | | | | | | | | | | | audio frames By using the frame counter (and the video time base) for audio pts we lose some timestamp precision but we ensure that video and audio coming from the same DV frame are always in sync. This patch also makes timestamps after seek consistent and it should also fix the timestamps when the audio clock is unlocked and have a completely indpendent clock source. (E.g. runs on fixed 48009 Hz which should have been exact 48000 Hz) Fixes out of sync timestamps in ticket #8762. Signed-off-by: Marton Balint <cus@passwd.hu>
* fate: add a test for HDR10+ metadata in HEVCMohammad Izadi2020-12-05
| | | | Signed-off-by: James Almer <jamrial@gmail.com>
* libavformat/mov.c: export vendor id as metadataThierry Foucu2020-12-05
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* avutil/opt: add AV_OPT_FLAG_DEPRECATED optionLimin Wang2020-12-05
| | | | Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
* fate: Convert the musepack8 test to an oneoff testMartin Storsjö2020-11-17
| | | | | | This fixes tests if built for x86 with x87 FPU. Signed-off-by: Martin Storsjö <martin@martin.st>
* aviobuf: Increase the default SHORT_SEEK_THRESHOLD to 32 KBMartin Storsjö2020-11-12
| | | | | | | | | | | | | | | | | | The previous threshold, 4 KB, maybe was reasonable when it was set (in 2010), but in today's settings and with typical network speeds and data sizes, it's pretty small. 32 KB probably is a more reasonable default now, regardless of input. This changes the test references for two seek tests. When using the normal seek function, which boils down to the lseek(2) function, a seek to an out of bounds position doesn't return an error, but that condition is only reported when doing the subsequent read (which returns EOF). When doing more seeks by fast forwarding, the fact that the seeked to destination is out of bounds is noticed and reported sooner in these cases. Signed-off-by: Martin Storsjö <martin@martin.st>
* avcodec/adpcm_ima_amv: use coded sample countZane van Iperen2020-11-09
| | | | Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
* avcodec/adpcm_ima_swf: fix frame size to 4096Zane van Iperen2020-11-07
| | | | | | | | | SWF File Format Specification, Version 19 says this is 1 raw sample + 4095 nibbles. https://www.adobe.com/content/dam/acom/en/devnet/pdf/swf-file-format-spec.pdf Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
* fate/filter-video: add 10bit test for unsharp filterLimin Wang2020-11-07
| | | | Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
* fate: Add test for Musepack SV8 decodingAndreas Rheinhardt2020-10-31
| | | | | | | | | | | | | | | While the FATE suite contains a sample file for Musepack 8, it did not use it to test the decoder; it is only used in the mpc8-demux test that tests the demuxer via streamcopy. Therefore this commit adds an actual encoder test. The test uses the framecrc output, because Musepack SV8 is an encoder that returns multiple frames for a single packet, so that timing information in the test output is valueable. Output seeking has been used in order to limit the size of the ref file as well as to test this codepath for the first time. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
* ffmpeg: move field order decision making to encoder initializationJan Ekström2020-10-29
| | | | | | | | | | | We now have the possibility of getting AVFrames here, and we should not touch the muxer's codecpar after writing the header. Results of FATE tests change as the MXF and Matroska muxers actually write down the field/frame coding type of a stream in their respective headers. Before this change, these values in codecpar would only be set after the muxer was initialized. Now, the information is also available for encoder and muxer initialization.
* ffmpeg: pass decoded or filtered AVFrame to output stream initializationJan Ekström2020-10-29
| | | | | | | | | | Additionally, reap the first rewards by being able to set the color related encoding values based on the passed AVFrame. The only tests that seem to have changed their results with this change seem to be the MXF tests. There, the muxer writes the limited/full range flag to the output container if the encoder is not set to "unspecified".
* lavf/url: fix relative url parsing when the query string or fragment has a colonruiquan.crq2020-10-28
| | | | | | | | | This disallows the usage of ? and # in libavformat specific scheme options (e.g. subfile,,start,32815239,end,0,,:video.ts) but this change was considered acceptable. Signed-off-by: ruiquan.crq <caihaoning83@gmail.com> Signed-off-by: Marton Balint <cus@passwd.hu>
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