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* aacenc: temporarily disable Mid/Side coding with multichannel filesRostislav Pehlivanov2016-02-131-0/+4
| | | | | | | | Results in dropping out in channels, usually on EIGHT_SHORT windows. Will be reenabled once the cause has been investigated and a fix has been made. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* aacenc: make a better estimate for the audio bitrate if not providedRostislav Pehlivanov2016-02-121-15/+31
| | | | | | | | | | | | | | Takes into account whether there's pairing and if there's an LFE channel. An SCE has more bits than CPE/2 since IS and M/S save quite a lot of bits when channels are paired. And most of the SCEs we have are in surround layouts which map it to the center channel, which usually carries all of the dialogue and compression artifacts there are easily audiable. Also refactors the init function a little bit and labels some parts of it. Fixes bug #5233 Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* avcodec/aacenc: Check all coefficients for finitenessMichael Niedermayer2016-01-201-11/+6
| | | | | | | | | | | | | This is needed as near infinite values on the input side result in only some output to be non finite. Also it may still be insufficient if subsequent computations overflow Fixes null pointer dereference Fixes: ae66c0f6c12ac1cd5c2c237031240f57/signal_sigsegv_2618c99_9516_6007026f2185a26d7afea895fbed6e38.ogg Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind Reviewed-by: Claudio Freire <klaussfreire@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* aacenc: remove FAAC-like coderRostislav Pehlivanov2016-01-201-4/+1
| | | | | | | Has been marked for removal for over a month and has not been improved or touched at all since it was implemented. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* aacenc: mark LTP mode as experimentalRostislav Pehlivanov2016-01-201-0/+3
| | | | | | | Too many crashes observed. Can't be helped until the autocorrelation function is massively checked for sanity. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* avcodec/aacenc: Check both channels for finitenessMichael Niedermayer2016-01-161-8/+8
| | | | | | | | Fixes null pointer dereference Fixes: 10412fc52ecc6eab40ed67f82ca7b372/signal_sigsegv_2618c99_2129_f808373959e46afb165593332799ffbc.aif Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* lavc/aacenc: use isfinite to simplify isnan/isinf logicGanesh Ajjanagadde2016-01-141-8/+9
| | | | | Reviewed-by: Claudio Freire <klaussfreire@gmail.com> Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
* avcodec/aacenc: Check for +-Inf tooMichael Niedermayer2016-01-131-9/+9
| | | | | | | | Fixes out of array read Fixes: 04442da73d935b776d2236282588d4f9/signal_sigsegv_2625a69_8790_ae85ffc889070663319b3417ede777b0.mov Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avcodec/aacenc: mark output as const as its not written toMichael Niedermayer2016-01-131-1/+1
| | | | Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* avcodec/aacenc: Fix NAN checkMichael Niedermayer2016-01-131-1/+9
| | | | | | | | | All MDCT outputs must be checked in case of 128point MDCTs Fixes: out of array read Fixes: 04442da73d935b776d2236282588d4f9/signal_sigsegv_2625a69_351_52ca6226eb83547a2d26e322ce84ed84.mov Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* AAC encoder: don't apply MS on special bandsClaudio Freire2016-01-131-2/+2
| | | | | | | | | Change the condition for application of the M/S transform to match that of the decoder. Namely, that no special coding books must be in use in either channel. While the condition ought to be equivalent to the current one when the invariant of is_mask is kept, matching the decoder's condition is safer and easier to maintain.
* acenc: remove deprecated avctx->frame_bits useRostislav Pehlivanov2015-12-181-6/+3
| | | | | | | | | The type of last_frame_pb_count was chosen to be an int since overflow is impossible (the spec says the maximum bits per frame is 6144 per channel and the encoder checks for that). Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com> Reviewed-by: Paul B Mahol <onemda@gmail.com>
* Merge commit '16216b713f9a21865cc07993961cf5d0ece24916'Hendrik Leppkes2015-12-181-0/+6
|\ | | | | | | | | | | | | * commit '16216b713f9a21865cc07993961cf5d0ece24916': lavc: Drop exporting 2-pass encoding stats Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
| * lavc: Drop exporting 2-pass encoding statsVittorio Giovara2015-12-071-4/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | These variables are coming from mpegvideoenc where are supposedly used as bit counters on various frame properties. However their use is unclear as they lack documentation, are available only from a very small subset of encoders, and they are hardly used in the wild. Also frame_bits in aacenc is employed in a similar way. Remove this functionality from AVCodecContex, these variable are mostly frame properties, and too few encoders support setting them with anything useful. Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* | aacenc: switch to using the RNG from libavutilRostislav Pehlivanov2015-12-141-1/+1
| | | | | | | | | | | | | | PSNR doesn't change as expected. The AAC spec doesn't really say anything about how exactly to generate noise. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: update max_sfb when num_swb changesAndreas Cadhalpun2015-12-081-0/+1
| | | | | | | | | | | | | | This fixes out-of-bounds reads in avoid_clipping. Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com> Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
* | Merge commit 'b805482b1fba1d82fbe47023a24c9261f18979b6'Hendrik Leppkes2015-12-081-1/+3
|\ \ | |/ | | | | | | | | | | * commit 'b805482b1fba1d82fbe47023a24c9261f18979b6': aac: Provide more information on the failure message Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
| * aac: Provide more information on the failure messageLuca Barbato2015-12-051-1/+3
| | | | | | | | Bug-Id: 761
* | aacenc: move the TNS search and filtering before PNSRostislav Pehlivanov2015-12-061-2/+2
| | | | | | | | | | | | | | | | The original plan was to have TNS use data from the PNS search to better tune itself to noise but this was never used nor necessary. This should slightly boost the PNS accuracy if TNS was used. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: fix aac_pred option triggering an errorRostislav Pehlivanov2015-12-051-1/+1
| | | | | | | | Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: remove the experimental flagRostislav Pehlivanov2015-12-051-2/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Thiss commit removes the experimental flag from the native AAC Encoder and thus makes it the default. After a lot of work, done by myself and Claudio Freire, the quality of this encoder rivals and surpasses libfdk_aac in some situations. The encoder had instability issues earlier which prevented it from having its experimental flag removed, however the last commits done by Claudio removed the last known source of instability and solved a lot of problems which were previously observed. The issues were caused by the various coding tools interfering with the scalefactor indices. Thus, with these problems solved, it should now be possible to declare this encoder as the default and recommend that the users should use it instead of others provided by external libraries, as it is both faster and has a subjectively higher quality with selected tracks. The encoder has still yet to be fine tuned for every possible audio file type like music or voice, so it is hoped that with the experimental flag removed the users should be able to provide feedback and make the encoder better than the alternatives for every type of audio and at every bitrate. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: mark coders other than twoloop as experimentalRostislav Pehlivanov2015-12-051-0/+2
| | | | | | | | | | | | | | ANMR has some interesting things coming up but is currently not in a shape fit for non-experimental usage. Same with "FAST". Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: mark the "faac"-like coder for removalRostislav Pehlivanov2015-12-051-0/+2
| | | | | | | | | | | | | | | | | | | | | | This coder produces a much lower quality audio than the rest, is much slower and is unstable. Hasn't been updated for a very long time as well, hence it is more appropriate to remove it since it also depends on a big burden of a code (the encode_window_bands_info function which is just as old, just as unstable and bad and in no way modifiable or fixable). Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | mips: rename mipsdspr1 to mipsdspVicente Olivert Riera2015-12-041-1/+1
| | | | | | | | | | Signed-off-by: Vicente Olivert Riera <Vincent.Riera@imgtec.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* | AAC encoder: improve SF range utilizationClaudio Freire2015-12-021-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch does 4 things, all of which interact and thus it woudln't be possible to commit them separately without causing either quality regressions or assertion failures. Fate comparison targets don't all reflect improvements in quality, yet listening tests show substantially improved quality and stability. 1. Increase SF range utilization. The spec requires SF delta values to be constrained within the range -60..60. The previous code was applying that range to the whole SF array and not only the deltas of consecutive values, because doing so requires smarter code: zeroing or otherwise skipping a band may invalidate lots of SF choices. This patch implements that logic to allow the coders to utilize the full dynamic range of scalefactors, increasing quality quite considerably, and fixing delta-SF-related assertion failures, since now the limitation is enforced rather than asserted. 2. PNS tweaks The previous modification makes big improvements in twoloop's efficiency, and every time that happens PNS logic needs to be tweaked accordingly to avoid it from stepping all over twoloop's decisions. This patch includes modifications of the sort. 3. Account for lowpass cutoff during PSY analysis The closer PSY's allocation is to final allocation the better the quality is, and given these modifications, twoloop is now very efficient at avoiding holes. Thus, to compute accurate thresholds, PSY needs to account for the lowpass applied implicitly during twoloop (by zeroing high bands). This patch makes twoloop set the cutoff in psymodel's context the first time it runs, and makes PSY account for it during threshold computation, making PE and threshold computations closer to the final allocation and thus achieving better subjective quality. 4. Tweaks to RC lambda tracking loop in relation to PNS Without this tweak some corner cases cause quality regressions. Basically, lambda needs to react faster to overall bitrate efficiency changes since now PNS can be quite successful in enforcing maximum bitrates, when PSY allocates too many bits to the lower bands, suppressing the signals RC logic uses to lower lambda in those cases and causing aggressive PNS. This tweak makes PNS much less aggressive, though it can still use some further tweaks. Also update MIPS specializations and adjust fuzz Also in lavc/mips/aacpsy_mips.h: remove trailing whitespace
* | aacenc: fix broken build with hardcoded tablesRostislav Pehlivanov2015-11-271-1/+6
| | | | | | | | | | | | | | | | ff_aac_tableinit is a macro in the case of hardcoded tables, so wrap that up in a function (similar to how the decoder template does it) and use that as the argument for ff_thread_once(). Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aac: temporarily un-share aac_table_init AVOnce variableRostislav Pehlivanov2015-11-271-0/+3
| | | | | | | | | | | | | | AAC-Fixed decoder segfaulted. This commit makes the aac encoder and decoder init the table twice in case of transcoding again. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: make threadsafeRostislav Pehlivanov2015-11-271-1/+3
| | | | | | | | | | | | | | | | Since the ff_aac_tableinit() can be called by both the encoder and the decoder (in case of transcoding) this commit shares the AVOnce variable to prevent this. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | AAC encoder: Fix application of M/S with PNSClaudio Freire2015-11-261-1/+7
| | | | | | | | | | | | | | | | When both M/S coding and PNS are enabled, scalefactors and coding books would be mistakenly clobbered when setting the M/S flag on PNS'd bands. The flag needs to be set to signal the generation of correlated noise, but the scalefactors, coefficients and the coding books need to be kept intact.
* | avcodec/aacenc: Fix "libavcodec/aacenc.c:540:13: warning: ISO C90 forbids ↵Michael Niedermayer2015-10-171-1/+1
| | | | | | | | | | | | mixed declarations and code" Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* | aacenc_tns: enable Temporal Noise Shaping by defaultRostislav Pehlivanov2015-10-171-1/+1
| | | | | | | | | | In light of the recent changes to the TNS system, it has been deemed worthy and robust enough to be turned on by default.
* | aacenc: partially revert previous commits to set options via a profileRostislav Pehlivanov2015-10-171-72/+38
| | | | | | | | | | | | It didn't work out because of the exceptions that needed to be made for the "-1" cases and was overall more confusing that just manually checking and setting options for each profile.
* | aacenc: add support for encoding files using Long Term PredictionRostislav Pehlivanov2015-10-171-0/+43
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Long Term Prediction allows for prediction of spectral coefficients via the previously decoded time-dependent samples. This feature works well with harmonic content 2 or more frames long, like speech, human or non-human, piano music or any constant tones at very low bitrates. It should be noted that the current coder is highly efficient and the rate control system is unable to encode files at extremely low bitrates (less than 14kbps seems to be impossible) so this extension isn't capable of optimum operation. Dramatic difference is observable with some types of audio and speech but for the most part the audiable differences are subtle. The spectrum looks better however so the encoder is able to harvest the additional bits that this feature provies, should the user choose to enable it. So it's best to enable this feature only if encoding at the absolutely lowest bitrate that the encoder is capable of.
* | aacenc: (re)enable Mid/Side coding by defaultRostislav Pehlivanov2015-10-171-1/+1
| | | | | | | | | | | | | | Apparently it was set to be enabled by default but after the profile commits it was reverted to be off by default because I didn't notice. Works well so (re)enable it.
* | aacenc: correctly zero prediction_used arrayRostislav Pehlivanov2015-10-171-1/+1
| | | | | | | | An oversight, probably because of copy-pasting the TNS line.
* | aacenc: slightly simplify and remove a redundant variableRostislav Pehlivanov2015-10-171-6/+8
| | | | | | | | Functionally identical, doesn't change anything.
* | aacenc: indicate that TNS is off by defaultRostislav Pehlivanov2015-10-171-2/+2
| | | | | | | | | | | | Doesn't change anything, just a slight clarification that under all profiles TNS is currently off. That'll be soon to change hopefully.
* | aacenc: shorten name of ff_aac_adjust_common_predictionRostislav Pehlivanov2015-10-121-2/+2
| | | | | | | | To keep it similar to the other functions which are all named *_pred.
* | aacenc: add support for changing options based on a profileRostislav Pehlivanov2015-10-121-48/+89
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This commit adds the ability for a profile to set the default options, as well as for the user to override such options by simply stating them in the command line while still keeping the same profile, as long as those options are still permitted by the profile. Example: setting the profile to aac_low (the default) will turn PNS and IS on. They can be disabled by -aac_pns 0 and -aac_is 0, respectively. Turning on -aac_pred 1 will cause the profile to be elevated to aac_main, as long as no options forbidding aac_main have been entered (like AAC-LTP, which will be pushed soon). A useful feature is that by setting the profile to mpeg2_aac_low, all MPEG4 features will be disabled and if the user tries to enable them then the program will exit with an error. This profile is signalled with the same bitstream as aac_low (MPEG4) but some devices and decoders will fail if any MPEG4 features have been enabled.
* | aacenc: add support for encoding 7.1 channel audioRostislav Pehlivanov2015-10-121-2/+3
| | | | | | | | | | | | | | This commit implements support for 7.1 channel audio. There's no more predefined bitstream channel mappings so going beyond 8 channels (and 7 channels exactly) will require programmable channel elements, which is already underway.
* | AAC encoder: memoize quantize_band_costClaudio Freire2015-10-121-0/+10
| | | | | | | | | | | | | | | | | | | | The bulk of calls to quantize_band_cost are replaced by a call to a version that memoizes, greatly improving performance, since during coefficient search there is a great deal of repeat work. Memoization cannot always be applied, so do this in a different function, and leave the original as-is.
* | AAC encoder: Extensive improvementsClaudio Freire2015-10-111-15/+23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This finalizes merging of the work in the patches in ticket #2686. Improvements to twoloop and RC logic are extensive. The non-exhaustive list of twoloop improvments includes: - Tweaks to distortion limits on the RD optimization phase of twoloop - Deeper search in twoloop - PNS information marking to let twoloop decide when to use it (turned out having the decision made separately wasn't working) - Tonal band detection and priorization - Better band energy conservation rules - Strict hole avoidance For rate control: - Use psymodel's bit allocation to allow proper use of the bit reservoir. Don't work against the bit reservoir by moving lambda in the opposite direction when psymodel decides to allocate more/less bits to a frame. - Retry the encode if the effective rate lies outside a reasonable margin of psymodel's allocation or the selected ABR. - Log average lambda at the end. Useful info for everyone, but especially for tuning of the various encoder constants that relate to lambda feedback. Psy: - Do not apply lowpass with a FIR filter, instead just let the coder zero bands above the cutoff. The FIR filter induces group delay, and while zeroing bands causes ripple, it's lost in the quantization noise. - Experimental VBR bit allocation code - Tweak automatic lowpass filter threshold to maximize audio bandwidth at all bitrates while still providing acceptable, stable quality. I/S: - Phase decision fixes. Unrelated to #2686, but the bugs only surfaced when the merge was finalized. Measure I/S band energy accounting for phase, and prevent I/S and M/S from being applied both. PNS: - Avoid marking short bands with PNS when they're part of a window group in which there's a large variation of energy from one window to the next. PNS can't preserve those and the effect is extremely noticeable. M/S: - Implement BMLD protection similar to the specified in ISO-IEC/13818:7-2003, Appendix C Section 6.1. Since M/S decision doesn't conform to section 6.1, a different method had to be implemented, but should provide equivalent protection. - Move the decision logic closer to the method specified in ISO-IEC/13818:7-2003, Appendix C Section 6.1. Specifically, make sure M/S needs less bits than dual stereo. - Don't apply M/S in bands that are using I/S Now, this of course needed adjustments in the compare targets and fuzz factors of the AAC encoder's fate tests, but if wondering why the targets go up (more distortion), consider the previous coder was using too many bits on LF content (far more than required by psy), and thus those signals will now be more distorted, not less. The extra distortion isn't audible though, I carried extensive ABX testing to make sure. A very similar patch was also extensively tested by Kamendo2 in the context of #2686.
* | AAC encoder: tweak rate-distortion logicClaudio Freire2015-09-231-22/+65
| | | | | | | | | | | | | | | | | | | | | | | | This patch modifies the encode frame function to retry encoding the frame when the resulting bit count is too far off target, but only adjusting lambda in small, incremental step. It also makes the logic more conservative - otherwise it will contend with bit reservoir-related variations in bit allocation, and result in artifacts when frame have to be truncated (usually at high bit rates transitioning from low complexity to high complexity).
* | avcodec/aacenc: use AV_OPT_TYPE_BOOLClément Bœsch2015-09-081-12/+4
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* | aacenc: copy PRNG from the decoderRostislav Pehlivanov2015-09-061-0/+1
| | | | | | | | | | | | Needed for the following PNS commits. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: reorder coding toolsRostislav Pehlivanov2015-09-021-63/+88
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This commit reorders the coding tools such that they're doing what the decoder does in reverse order. The very first thing the decoder does is to decode M/S stereo if that's signalled, then prediction, IS, and finally TNS and PNS in another function. adjust_frame_information()'s application of IS and M/S was taken out into two separate functions since prediction doesn't expect to get the raw coefficients but rathe the coefficients at that part of the encoding process. The results show a much better PSNR when any combination of Intensity Stereo, Mid/Side stereo and Prediction is used, which is a sign of an increased encoder efficiency as well as the fact that the decoder gets what it expects. Otherwise, with only IS, PNS or prediction there are neither regressions nor improvements except in the case of IS, which now by itself (or with PNS) is less prone to artifacts. Enabling M/S (using stereo_mode) as well will also reduce stereo artifacts induced by IS, so in the very near future M/S may be enabled by default. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: disable bandtype modifying extensions when coder != twoloopRostislav Pehlivanov2015-09-011-0/+5
| | | | | | | | | | | | | | | | | | If the selected coder isn't twoloop, this commit temporarily disables IS and PNS. The problem is in the encode_window_bands_info() being confused and setting invalid band_types for non-marked (normal) bands. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: Enable Intensity Stereo by defaultRostislav Pehlivanov2015-09-011-1/+1
| | | | | | | | | | | | | | | | | | | | | | Since the changes made a few week ago (which were done more than a month ago) the quality and stability of intensity stereo has been notably good. There were some requests and wishes to have in on by default and therefore it has been enabled. Should any regressions arise changes will be made to preferably keep it operating rather than just disabling it by default again. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: Enable Perceptual Noise Substitution by defaultRostislav Pehlivanov2015-09-011-1/+1
| | | | | | | | | | | | | | | | It has been in the current encoder in its current implementation for quite some time now, so enable it by default. Will increase quality at all bitrates, especially at low ones. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
* | aacenc: reorder resetting of cpe->common_windowRostislav Pehlivanov2015-09-011-1/+1
| | | | | | | | | | | | | | Purely a cosmetic change, most of the zeroing of encoder resources should happen at the top of the main loop. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
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