- 17 Oct, 2015 1 commit
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Rostislav Pehlivanov authored
Changes: - strongly prefer dual filters to a single filter - less strict about using 2 filters w.r.t. energy - scrap the usage of threshold and spread, useless - use odd-shaped windows to set the filter direction - use 4 bits instead of 3 bits for short windows - simplify and reduce the main loop to a single level - add stricter regulations for short windows All of this now makes the TNS implementation operate as good as it can and it definitely shows. The frequency thresholds are now even better defined by looking at the spectrals and the overall sound has been improved at the price of just a few bits that are well worth it.
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- 12 Sep, 2015 2 commits
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Rostislav Pehlivanov authored
Since TNS was fixed with the recent commits retweak the values so it's more frequently used. Still not enabled by default yet, though it's possible that it will be made enabled by default in the near future. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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Rostislav Pehlivanov authored
This finally (and again) gets rid of basically everything the specifications say about how TNS should be done. The main problem used to be that a single filter was used for all coefficients which despite being explicitly recommended by the specifications usually sounds wrong, therefore it's a corner case in the current TNS implementation. This commit also changes the coefficient bit size, as apparently it's better to use lower precision in case the windows are eight short. This is apparently what fdk_aac uses, looking at the bit stream and makes sense. Also the order when 8 SHORT windows happen is important as 7 was too much and according to PSNR was worse while 5 is just about correct. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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- 05 Sep, 2015 1 commit
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Rostislav Pehlivanov authored
This commit improves the TNS implementation to the point where it's actually usable and very rarely results in nastyness (in all bitrates except extremely low bitrates it's increasing the quality and prevents some distortions from the coder being audiable). Also adds a double filter support which is only used if the energy difference between the top and bottom of the SFBs is above the thresholds defined in the header file. Looking at the bitstream that fdk_aac generates it sometimes used a double filter despite the specs stating that a single filter should be enough for almost all cases and purposes. Unlike FAAC or fdk_aac we sometimes use a reverse filter in case the energy difference isn't enought to use a double filter. This actually works better. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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- 01 Sep, 2015 1 commit
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Rostislav Pehlivanov authored
This commit reworks the TNS implementation to a hybrid between what the specifications say, what the decoder does and what's the best thing to do. The filter application function was copied from the decoder and modified such that it applies the inverse AR filter to the coefficients. The LPC coefficients themselves are fed into the same quantization expression that the specifications say should be used however further processing is not done, instead they're converted to the form that the decoder expects them to be in and are sent off to the compute_lpc_coeffs function exactly the way the decoder does. This function does all conversions and will return the exact coefficients that the decoder will generate, which are then applied to the coefficients. Having the exact same coefficients on both the encoder and decoder is a must since otherwise the entire sfb's over which the filter is applied will be attenuated. Despite this major rework, TNS might not work fine on some audio types at very low bitrates (e.g. sub 90kbps) as it can attenuate some coefficients too much. Users are advised to experiment with TNS at higher bitrates if they wish to use this tool or simply wait for the implementation to be improved. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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- 29 Aug, 2015 1 commit
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Rostislav Pehlivanov authored
This commit abandons the way the specifications state to quantize the coefficients, makes use of the new LPC float functions and is much better. The original way of converting non-normalized float samples to int32_t which out LPC system expects was wrong and it was wrong to assume the coefficients that are generated are also valid. It was essentially a full garbage-in, garbage-out system and it definitely shows when looking at spectrals and listening. The high frequencies were very overattenuated. The new LPC function performs the analysis directly. The specifications state to quantize the coefficients into four bit index values using an asin() function which of course had to have ugly ternary operators because the function turns negative if the coefficients are negative which when encoding causes invalid bitstream to get generated. This deviates from this by using the direct TNS tables, which are fairly small since you only have 4 bits at most for index values. The LPC values are directly quantized against the tables and are then used to perform filtering after the requantization, which simply fetches the array values. The end result is that TNS works much better now and doesn't attenuate anything but the actual signal, e.g. TNS removes quantization errors and does it's job correctly now. It might be enabled by default soon since it doesn't hurt and helps reduce nastyness at low bitrates. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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- 22 Aug, 2015 2 commits
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Timothy Gu authored
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Timothy Gu authored
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com> Reviewed-by: Ganesh Ajjanagadde <gajjanag@mit.edu>
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- 21 Aug, 2015 1 commit
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Rostislav Pehlivanov authored
This commit implements temporal noise shaping support in the encoder, along with an -aac_tns option to toggle it on or off (off by default for now). TNS will increase audio quality and reduce quantization noise by applying a multitap FIR filter across allowed coefficients and transmit side information to the decoder so it could create an inverse filter. Users are encouraged to test the new functionality by enabling -aac_tns 1 during encoding. No major bugs are observable at this time so after a while if no new problems appear and if the current implementation is deemed of high enough quality and stability it will be enabled by default, possibly at the same time the encoder has its experimental flag removed and becomes the standard aac encoder in ffmpeg. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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- 18 Oct, 2012 1 commit
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Martin Storsjö authored
This requires the makedef perl script by Derek, from the c89-to-c99 repo. That scripts produces a .def file, listing the symbols to be exported, based on the gcc version scripts and the built object files. To properly load non-function symbols from DLL files, the data symbol declarations need to have the attribute __declspec(dllimport) when building the calling code. (On mingw, the linker can fix this up automatically, which is why it has not been an issue so far. If this attribute is omitted, linking actually succeeds, but reads from the table will not produce the desired results at runtime.) MSVC seems to manage to link DLLs (and run properly) even if this attribute is present while building the library itself (which normally isn't recommended) - other object files in the same library manage to link to the symbol (with a small warning at link time, like "warning LNK4049: locally defined symbol _avpriv_mpa_bitrate_tab imported" - it doesn't seem to be possible to squelch this warning), and the definition of the tables themselves produce a warning that can be squelched ("warning C4273: 'avpriv_mpa_bitrate_tab' : inconsistent dll linkage, see previous definition of 'avpriv_mpa_bitrate_tab'). In this setup, mingw isn't able to link object files that refer to data symbols with __declspec(dllimport) without those symbols actually being linked via a DLL (linking avcodec.dll ends up with errors like "undefined reference to `__imp__avpriv_mpa_freq_tab'"). The dllimport declspec isn't needed at all in mingw, so we simply choose not to declare it for other compilers than MSVC that requires it. (If ICL support later requires it, the condition can be extended later to include both of them.) This also implies that code that is built to link to a certain library as a DLL can't link to the same library as a static library. Therefore, we only allow building either static or shared but not both at the same time. (That is, static libraries as such can be, and actually are, built - this is used for linking the test tools to internal symbols in the libraries - but e.g. libavformat built to link to libavcodec as a DLL cannot link statically to libavcodec.) Also, linking to DLLs is slightly different from linking to shared libraries on other platforms. DLLs use a thing called import libraries, which is basically a stub library allowing the linker to know which symbols exist in the DLL and what name the DLL will have at runtime. In mingw/gcc, the import library is usually named libfoo.dll.a, which goes next to a static library named libfoo.a. This allows gcc to pick the dynamic one, if available, from the normal -lfoo switches, just as it does for libfoo.a vs libfoo.so on Unix. On MSVC however, you need to literally specify the name of the import library instead of the static library. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 24 Aug, 2012 1 commit
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Mans Rullgard authored
This table is used only by mpegaudiodsp and mpegaudioenc. Separating it allows dropping some dependencies from mpc[78] and qdm2. Signed-off-by: Mans Rullgard <mans@mansr.com>
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- 20 Oct, 2011 1 commit
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Anton Khirnov authored
Specifically, ff_mpa_freq_tab, ff_mpa_bitrate_tab, ff_mpa_decode_header, ff_mpegaudio_decode_header.
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- 20 May, 2011 1 commit
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Mans Rullgard authored
Signed-off-by: Mans Rullgard <mans@mansr.com>
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- 19 Mar, 2011 1 commit
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Mans Rullgard authored
Signed-off-by: Mans Rullgard <mans@mansr.com>
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- 20 Apr, 2010 1 commit
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Diego Biurrun authored
Passing an explicit filename to this command is only necessary if the documentation in the @file block refers to a file different from the one the block resides in. Originally committed as revision 22921 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 01 Feb, 2009 1 commit
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Diego Biurrun authored
Otherwise doxygen complains about ambiguous filenames when files exist under the same name in different subdirectories. Originally committed as revision 16912 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 31 Aug, 2008 1 commit
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Stefano Sabatini authored
Consistently apply this rule: the guard name is obtained from the filename by stripping the leading "lib", converting '/' and '.' to '_' and uppercasing the resulting name. Guard names in the root directory have to be prefixed by "FFMPEG_". Originally committed as revision 15120 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 24 Jun, 2008 1 commit
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Stefan Gehrer authored
Originally committed as revision 13947 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 09 May, 2008 1 commit
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Diego Biurrun authored
Originally committed as revision 13098 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 08 Nov, 2007 1 commit
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Martin Lindhe authored
when DEBUG is defined. Patch by Martin Lindhe: martin startwars org Originally committed as revision 10955 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 17 Oct, 2007 1 commit
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Diego Biurrun authored
Originally committed as revision 10765 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 19 May, 2007 2 commits
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Aurelien Jacobs authored
Originally committed as revision 9061 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Aurelien Jacobs authored
Originally committed as revision 9060 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 27 Feb, 2007 1 commit
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Diego Biurrun authored
Originally committed as revision 8149 to svn://svn.ffmpeg.org/ffmpeg/trunk
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- 25 Jul, 2002 1 commit
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Fabrice Bellard authored
Originally committed as revision 809 to svn://svn.ffmpeg.org/ffmpeg/trunk
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