{"push_id":776743853,"size":58,"distinct_size":58,"ref":"refs/heads/master","head":"63fb5a6aefb4223334001fd2c0d82a5e22e3b528","before":"53bf32fa4270eb60f3bf25b5bd3d1f97381d4661","commits":[{"sha":"629d4c5b4deee08bf3a4f3ab45fd4f8b76d7aff3","author":{"email":"1ed1015336f6a462e9365b18ca61d9eb32e77fa2@gmail.com","name":"Timothy Gu"},"message":"avconv_opt: Add missing comma\n\nSigned-off-by: Luca Barbato <lu_zero@gentoo.org>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/629d4c5b4deee08bf3a4f3ab45fd4f8b76d7aff3"},{"sha":"c23999be134bde0a0554261a9043be7dbc01de0c","author":{"email":"1ed1015336f6a462e9365b18ca61d9eb32e77fa2@gmail.com","name":"Timothy Gu"},"message":"avconv_opt: Add an option that lists all supported hwaccels\n\nSigned-off-by: Luca Barbato <lu_zero@gentoo.org>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/c23999be134bde0a0554261a9043be7dbc01de0c"},{"sha":"b1abd2aaf91be249f24cb74db9c205d9e4ca9da6","author":{"email":"df7b252cf76d4b413544c9b5ebe9222d5feffae0@gentoo.org","name":"Luca Barbato"},"message":"vf_scale: Add an option to pass the scaler params\n\nReported-By: zehan@magicpony.technology","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/b1abd2aaf91be249f24cb74db9c205d9e4ca9da6"},{"sha":"c457bdebe7af643b380aa0f6add3cb4335d218dc","author":{"email":"a076b11c2cc63ccdc903dbe182e1eecc3cbbc094@gramner.com","name":"Henrik Gramner"},"message":"checkasm: Fix floating point arguments on 64-bit Windows\n\nSigned-off-by: Anton Khirnov <anton@khirnov.net>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/c457bdebe7af643b380aa0f6add3cb4335d218dc"},{"sha":"9f2fa95bd8e85aab31dcb24977081eff15d92359","author":{"email":"7d5360c84cae42bfaaea1fae3d03b3a6bbc5fab6@gmail.com","name":"Paul B Mahol"},"message":"avfilter/vf_histogram: 9 and 10 bit depth support\n\nSigned-off-by: Paul B Mahol <onemda@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/9f2fa95bd8e85aab31dcb24977081eff15d92359"},{"sha":"1e50f953fac7fb3af907a24f9fc301fa80f441b4","author":{"email":"c2e2ad641b78da5d5e231b38d6c2530ebaa29faa@overt.org","name":"Philip Langdale"},"message":"Changelog: Add VDPAU HEVC to the list\n\nSigned-off-by: Philip Langdale <philipl@overt.org>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/1e50f953fac7fb3af907a24f9fc301fa80f441b4"},{"sha":"2fb593dcb90c157a183e9e01e42405fcca73a8b8","author":{"email":"90ccf6571a312491d0ea6b0e16399e672356a479@gmail.com","name":"Ronald S. Bultje"},"message":"Put remaining pieces of CODEC_FLAG_EMU_EDGE under FF_API_EMU_EDGE.\n\nThe amv one probably looks suspicious, but since it's an intra-only\ncodec, I couldn't possibly imagine what it would use the edge for,\nand the vsynth fate result doesn't change, so it's probably OK.","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/2fb593dcb90c157a183e9e01e42405fcca73a8b8"},{"sha":"7e12a54251148f8d2756161bcf09f631a6c144db","author":{"email":"90ccf6571a312491d0ea6b0e16399e672356a479@gmail.com","name":"Ronald S. Bultje"},"message":"ffserver: use -b instead of -ab for setting audio bitrate.","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/7e12a54251148f8d2756161bcf09f631a6c144db"},{"sha":"777df1ff746ea8a264971956cb09f3e199342151","author":{"email":"7d5360c84cae42bfaaea1fae3d03b3a6bbc5fab6@gmail.com","name":"Paul B Mahol"},"message":"avfilter/vf_dejudder: use the name 's' for the pointer to the private context\n\nThis is shorter and consistent across filters.\n\nSigned-off-by: Paul B Mahol <onemda@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/777df1ff746ea8a264971956cb09f3e199342151"},{"sha":"628a73f8f3768513fa6152c98d54842cf2ae1aad","author":{"email":"17b9e1c64588c7fa6419b4d29dc1f4426279ba01@niedermayer.cc","name":"Michael Niedermayer"},"message":"ffmpeg: force 128k default audio bitrate if nothing is specified and there is no specific default\n\nThis prevents breaking existing command lines in case the \"ab\" default is removed from libavcodec\n\nSigned-off-by: Michael Niedermayer <michael@niedermayer.cc>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/628a73f8f3768513fa6152c98d54842cf2ae1aad"},{"sha":"91f1115a0e027074c90bac6e57c2d0f4fe9efe8c","author":{"email":"c2e2ad641b78da5d5e231b38d6c2530ebaa29faa@overt.org","name":"Philip Langdale"},"message":"avcodec/vc1dec: Re-order init to avoid initting hwaccel too early\n\nAt least for vdpau, the hwaccel init code tries to check the video\nprofile and ensure that there is a matching vdpau profile available.\n\nIf it can't find a match, it will fail to initialise.\n\nIn the case of wmv3/vc1, I observed initialisation to fail all the\ntime. It turns out that this is due to the hwaccel being initialised\nvery early in the codec init, before the profile has been extracted\nand set.\n\nConceptually, it's a simple fix to reorder the init code, but it gets\nmessy really fast because ff_get_format(), which is what implicitly\ntrigger hwaccel init, is called multiple times through various shared\ninit calls from h263, etc. It's incredibly hard to prove to my own\nsatisfaction that it's safe to move the vc1 specific init code\nahead of this generic code, but all the vc1 fate tests pass, and I've\nvisually inspected a couple of samples and things seem correct.\n\nSigned-off-by: Philip Langdale <philipl@overt.org>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/91f1115a0e027074c90bac6e57c2d0f4fe9efe8c"},{"sha":"f55cc579115e4bcfe26eb05ecaa1c823343e9e1e","author":{"email":"8bc83ecdb1b2231045bfd7961620976748f2bd2d@gmail.com","name":"Rostislav Pehlivanov"},"message":"aac: move the TNS tables from aacdectab to the shared aactab\n\nThis commit simply moves the TNS tables to a more appropriate\naactab.h since then they can be accessed by both the decoder\nand encoder.\n\nThe encoder _shouldn't_ normally need the tables since the\nspecs describe a specific quantization process, but the exact\nreason for this can be seen in the TNS commit following.\n\nSigned-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/f55cc579115e4bcfe26eb05ecaa1c823343e9e1e"},{"sha":"20962b567b4d1fe791ca4b537121f13f6e0f5fa0","author":{"email":"8bc83ecdb1b2231045bfd7961620976748f2bd2d@gmail.com","name":"Rostislav Pehlivanov"},"message":"lpc: create a simplified Levinson-Durbin LPC handling float samples\n\nThis commit simply duplicates the functionality of ff_lpc_calc_coefs()\nfor the case of a Levinson-Durbin LPC with the only difference being\nthat floating point samples are accepted and the resulting coefficients\nare raw and unquantized.\nThe motivation behind doing this is the fact that the AAC encoder\nrequires LPC in TNS and LTP and converting non-normalized floating\npoint coefficients to int32_t using SWR and again back for the LPC\ncoefficients was very impractical.\nThe current LPC interfaces were designed for int32_t in mind possibly\nbecause FLAC and ALAC use this type for most internal operations.\nThe mathematics in case of floats remains of course identical.\n\nSigned-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/20962b567b4d1fe791ca4b537121f13f6e0f5fa0"},{"sha":"949a4892fa1dd6da028229cb1674340542fc9bc4","author":{"email":"8bc83ecdb1b2231045bfd7961620976748f2bd2d@gmail.com","name":"Rostislav Pehlivanov"},"message":"aacenc: change FF_PROFILE_UNKNOWN to AAC-Main if prediction is enabled\n\nThis was missed when the original commits were done. FF_PROFILE_UNKNOWN\nis what's in avctx->profile when no audio profile is specified.\n\nSigned-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/949a4892fa1dd6da028229cb1674340542fc9bc4"},{"sha":"44ddee945a2e8cfc1b7074de6e35595ed41da4e4","author":{"email":"8bc83ecdb1b2231045bfd7961620976748f2bd2d@gmail.com","name":"Rostislav Pehlivanov"},"message":"aacenc_pred: rework the way prediction is done\n\nThis commit completely alters the algorithm of prediction.\nThe original commit which introduced prediction was completely\nincorrect to even remotely care about what the actual coefficients\ncontain or whether any options were enabled. Not my actual fault.\n\nThis commit treats prediction the way the decoder does and expects\nto do: like lossy encryption. Everything related to prediction now\nhappens at the very end but just before quantization and encoding\nof coefficients. On the decoder side, prediction happens before\nanything has had a chance to even access the coefficients.\n\nAlso the original implementation had problems because it actually\ntouched the band_type of special bands which already had their\nscalefactor indices marked and it's a wonder the asserion wasn't\ntriggered when transmitting those.\n\nOverall, this now drastically increases audio quality and you should\nthink about enabling it if you don't plan on playing anything encoded\non really old low power ultra-embedded devices since they might not\nsupport decoding of prediction or AAC-Main. Though the specifications\nwere written ages ago and as times change so do the FLOPS.\n\nSigned-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/44ddee945a2e8cfc1b7074de6e35595ed41da4e4"},{"sha":"1cd5daee20602e1fa271bd38b3aa47c9bc30ac67","author":{"email":"8bc83ecdb1b2231045bfd7961620976748f2bd2d@gmail.com","name":"Rostislav Pehlivanov"},"message":"aac: remove now-unused redundant array\n\nThis commit removes the array which was made redundant with\nthe last commit. The current prediction system gets the\nquantization error directly (and without the single-frame delay)\nin the search_for_pred function.\n\nSigned-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/1cd5daee20602e1fa271bd38b3aa47c9bc30ac67"},{"sha":"f20b67173ca6a05b8c3dee02dad3b7243b96292b","author":{"email":"8bc83ecdb1b2231045bfd7961620976748f2bd2d@gmail.com","name":"Rostislav Pehlivanov"},"message":"aacenc_tns: rework the way coefficients are calculated\n\nThis commit abandons the way the specifications state to\nquantize the coefficients, makes use of the new LPC float\nfunctions and is much better.\n\nThe original way of converting non-normalized float samples\nto int32_t which out LPC system expects was wrong and it was\nwrong to assume the coefficients that are generated are also\nvalid. It was essentially a full garbage-in, garbage-out\nsystem and it definitely shows when looking at spectrals\nand listening. The high frequencies were very overattenuated.\nThe new LPC function performs the analysis directly.\n\nThe specifications state to quantize the coefficients into\nfour bit index values using an asin() function which of course\nhad to have ugly ternary operators because the function turns\nnegative if the coefficients are negative which when encoding\ncauses invalid bitstream to get generated.\n\nThis deviates from this by using the direct TNS tables, which\nare fairly small since you only have 4 bits at most for index\nvalues. The LPC values are directly quantized against the tables\nand are then used to perform filtering after the requantization,\nwhich simply fetches the array values.\n\nThe end result is that TNS works much better now and doesn't\nattenuate anything but the actual signal, e.g. TNS removes\nquantization errors and does it's job correctly now.\n\nIt might be enabled by default soon since it doesn't hurt and\nhelps reduce nastyness at low bitrates.\n\nSigned-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/f20b67173ca6a05b8c3dee02dad3b7243b96292b"},{"sha":"f04d86c16a5dd884114653441101c42d98fb28f7","author":{"email":"8bc83ecdb1b2231045bfd7961620976748f2bd2d@gmail.com","name":"Rostislav Pehlivanov"},"message":"aacenc: remove TNS from the todo list\n\nPulses are already on the way so expect to see the list\ngone in the close future.\n\nTNS is already of sufficiently high quality to be enabled\nby default (but isn't yet, so you too can help by testing!).\n\nSigned-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/f04d86c16a5dd884114653441101c42d98fb28f7"},{"sha":"fe12ba6f30127a78effdc7cdc8ec92f0830d0cf0","author":{"email":"8bc83ecdb1b2231045bfd7961620976748f2bd2d@gmail.com","name":"Rostislav Pehlivanov"},"message":"fate: reenable TNS test\n\nHopefully without errors like last time, but I'm prepared.\n\nSigned-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/fe12ba6f30127a78effdc7cdc8ec92f0830d0cf0"},{"sha":"4ff897a5364f85de051ca81058554454f3ad5d62","author":{"email":"8bc83ecdb1b2231045bfd7961620976748f2bd2d@gmail.com","name":"Rostislav Pehlivanov"},"message":"fate: add a test for encoding AAC-Main prediction\n\nThis commit introduces a test for AAC-Main prediction\nwhich was just reworked in this series of commits.\n\nSigned-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>","distinct":true,"url":"https://api.github.com/repos/KevinJW/FFmpeg/commits/4ff897a5364f85de051ca81058554454f3ad5d62"}]}