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FFmpeg/libavcodec/aacenc.h

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Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
/*
* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AACENC_H
#define AVCODEC_AACENC_H
Add a float DSP framework to libavutil Move vector_fmul() from DSPContext to AVFloatDSPContext.
2012-05-21 19:58:41 +03:00
#include "libavutil/float_dsp.h"
aacenc: switch to using the RNG from libavutil 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>
2015-12-14 20:53:09 +02:00
#include "libavutil/lfg.h"
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
#include "avcodec.h"
#include "put_bits.h"
#include "aac.h"
aacenc: use AVCodec.encode2()
2012-01-28 20:28:01 +03:00
#include "audio_frame_queue.h"
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
#include "psymodel.h"
aacenc: create and initialize an LTP context This commit only creates and initializes an LTP context which is needed for upcoming commits (TNS). Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-08-21 19:43:09 +02:00
#include "lpc.h"
avcodec/aacenc: use enum for aac coder. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-12 18:25:30 +03:00
typedef enum AACCoder {
aacenc: remove FAAC-like coder 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>
2016-01-20 18:56:53 +02:00
AAC_CODER_ANMR = 0,
avcodec/aacenc: use enum for aac coder. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-12 18:25:30 +03:00
AAC_CODER_TWOLOOP,
AAC_CODER_FAST,
aacenc: add AAC_CODER_(FAAC|ANMR|etc.) macros Signed-off-by: Timothy Gu <timothygu99@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-11 06:23:32 +03:00
avcodec/aacenc: use enum for aac coder. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-09-12 18:25:30 +03:00
AAC_CODER_NB,
}AACCoder;
aacenc: add AAC_CODER_NB Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-27 23:19:07 +03:00
aacenc: Add stereo_mode option. ms_off is the default, until Mid/Side is no longer buggy. Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-06-01 08:38:09 +03:00
typedef struct AACEncOptions {
aacenc: add support for changing options based on a profile 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.
2015-10-12 17:50:10 +02:00
int coder;
aaccoder: Implement Perceptual Noise Substitution for AAC This commit implements the perceptual noise substitution AAC extension. This is a proof of concept implementation, and as such, is not enabled by default. This is the fourth revision of this patch, made after some problems were noted out. Any changes made since the previous revisions have been indicated. In order to extend the encoder to use an additional codebook, the array holding each codebook has been modified with two additional entries - 13 for the NOISE_BT codebook and 12 which has a placeholder function. The cost system was modified to skip the 12th entry using an array to map the input and outputs it has. It also does not accept using the 13th codebook for any band which is not marked as containing noise, thereby restricting its ability to arbitrarily choose it for bands. The use of arrays allows the system to be easily extended to allow for intensity stereo encoding, which uses additional codebooks. The 12th entry in the codebook function array points to a function which stops the execution of the program by calling an assert with an always 'false' argument. It was pointed out in an email discussion with Claudio Freire that having a 'NULL' entry can result in unexpected behaviour and could be used as a security hole. There is no danger of this function being called during encoding due to the codebook maps introduced. Another change from version 1 of the patch is the addition of an argument to the encoder, '-aac_pns' to enable and disable the PNS. This currently defaults to disable the PNS, as it is experimental. The switch will be removed in the future, when the algorithm to select noise bands has been improved. The current algorithm simply compares the energy to the threshold (multiplied by a constant) to determine noise, however the FFPsyBand structure contains other useful figures to determine which bands carry noise more accurately. Some of the sample files provided triggered an assertion when the parameter to tune the threshold was set to a value of '2.2'. Claudio Freire reported the problem's source could be in the range of the scalefactor indices for noise and advised to measure the minimal index and clip anything above the maximum allowed value. This has been implemented and all the files which used to trigger the asserion now encode without error. The third revision of the problem also removes unneded variabes and comparisons. All of them were redundant and were of little use for when the PNS implementation would be extended. The fourth revision moved the clipping of the noise scalefactors outside the second loop of the two-loop algorithm in order to prevent their redundant calculations. Also, freq_mult has been changed to a float variable due to the fact that rounding errors can prove to be a problem at low frequencies. Considerations were taken whether the entire expression could be evaluated inside the expression , but in the end it was decided that it would be for the best if just the type of the variable were to change. Claudio Freire reported the two problems. There is no change of functionality (except for low sampling frequencies) so the spectral demonstrations at the end of this commit's message were not updated. Finally, the way energy values are converted to scalefactor indices has changed since the first commit, as per the suggestion of Claudio Freire. This may still have some drawbacks, but unlike the first commit it works without having redundant offsets and outputs what the decoder expects to have, in terms of the ranges of the scalefactor indices. Some spectral comparisons: https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Original.png (original), https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_NO.png (encoded without PNS), https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS1.2.png (encoded with PNS, const = 1.2), https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Difference1.png (spectral difference). The constant is the value which multiplies the threshold when it gets compared to the energy, larger values means more noise will be substituded by PNS values. Example when const = 2.2: https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_2.2.png Reviewed-by: Claudio Freire <klaussfreire@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-04-15 13:18:42 +02:00
int pns;
aacenc_tns: implement temporal noise shaping 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>
2015-08-21 20:27:38 +02:00
int tns;
aacenc: add support for encoding files using Long Term Prediction 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.
2015-10-17 03:22:51 +02:00
int ltp;
aacenc: do not reject AAC-Main profile This commit permits for the use of the Main profile in encoding. The functionality of that profile will be added in the commits following. By itself, this commit does not alter anything. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-08-21 20:20:22 +02:00
int pred;
aacenc: add support for changing options based on a profile 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.
2015-10-12 17:50:10 +02:00
int mid_side;
aacenc: implement Intensity Stereo encoding support This commit implements intensity stereo coding support to the native aac encoder. This is a way to increase the efficiency of the encoder by zeroing the right channel's spectral coefficients (in a channel pair) and rederiving them in the decoder using information from the scalefactor indices of special band types. This commit confomrs to the official ISO 13818-7 specifications, although due to their ambiguity certain deviations have been taken to ensure maximum sound quality. This commit has been extensively tested and has shown to not result in audiable audio artifacts unless in extreme cases. This commit also adds an option, aac_is, which has the value of 0 by default. Intensity Stereo is part of the scalable aac profile and is thus non-default. The way IS coding works is that it rederives the right channel's spectral coefficients from the left channel via the scalefactor index values left in the right channel. Since an entire band's spectral coefficients do not need to be coded, the encoder's efficiency jumps up and it unzeroes some high frequency values which it previously did not have enough bits to encode. That way less information is lost than the information lost by rederiving the spectral coefficients with some error. This is why the filesize of files encoded with IS do not decrease significantly. Users wishing that IS coding should reduce filesize are expected to reduce their encoding bitrates appropriately. This is V2 of the commit. The old version did not mark ms_mask as 0 since M/S and IS coding are incompactible, which resulted in distortions with M/S coding enabled. This version also improves phase detection by measuring it for every spectral coefficient in the band and using a simple majority rule to determine whether the coefficients are in or out of phase. Also, the energy values per spectral coefficient were changed as to reflect the official specifications. Reviewed-by: Claudio Freire <klaussfreire@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-07-02 20:13:07 +02:00
int intensity_stereo;
aacenc: Add stereo_mode option. ms_off is the default, until Mid/Side is no longer buggy. Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-06-01 08:38:09 +03:00
} AACEncOptions;
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
struct AACEncContext;
Cosmetics: Pretty print the AAC encoder. Originally committed as revision 19376 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:36:45 +03:00
typedef struct AACCoefficientsEncoder {
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
SingleChannelElement *sce, const float lambda);
void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda);
aaccoder: move the quantization functions to a separate file This commit moves the quantizer to a separate header file. This allows the quantizer to be used from a separate files outside of aaccoder without having to put another function pointer and will result in a slight speedup as the compiler can do more optimizations. This is required for commits following. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-08-21 19:53:14 +02:00
void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size,
AAC Encoder: clipping avoidance Avoid clipping due to quantization noise to produce audible artifacts, by detecting near-clipping signals and both attenuating them a little and encoding escape-encoded bands (usually the loudest) rounding towards zero instead of nearest, which tends to decrease overall energy and thus clipping. Currently fate tests measure numerical error so this change makes tests using asynth (which are near clipping) report higher error not less, because of window attenuation. Yet, they sound better, not worse (albeit subtle, other samples aren't subtle at all). Only measuring psychoacoustically weighted error would make for a representative test, so that will be left for a future patch. Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2015-07-21 03:53:24 +02:00
int scale_idx, int cb, const float lambda, int rtz);
aacenc_tns: implement temporal noise shaping 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>
2015-08-21 20:27:38 +02:00
void (*encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce);
aacenc: add support for encoding files using Long Term Prediction 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.
2015-10-17 03:22:51 +02:00
void (*encode_ltp_info)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
aacenc: implement the complete AAC-Main profile This commit finalizes AAC-Main profile encoding support by implementing all mandatory and optional tools available in the specifications and current decoders. The AAC-Main profile reqires that prediction support be present (although decoders don't require it to be enabled) for an encoder to be deemed capable of AAC-Main encoding, as well as TNS, PNS and IS, all of which were implemented with previous commits or earlier of this year. Users are encouraged to test the new functionality using either -profile:a aac_main or -aac_pred 1, the former of which will enable the prediction option by default and the latter will change the profile to AAC-Main. No other options shall be changed by enabling either, it's currently up to the users to decide what's best. The current implementation works best using M/S and/or IS, so users are also welcome to enable both options and any other options (TNS, PNS) for maximum quality. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-08-21 20:38:05 +02:00
void (*encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
aacenc: shorten name of ff_aac_adjust_common_prediction To keep it similar to the other functions which are all named *_pred.
2015-10-13 00:33:07 +02:00
void (*adjust_common_pred)(struct AACEncContext *s, ChannelElement *cpe);
aacenc: add support for encoding files using Long Term Prediction 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.
2015-10-17 03:22:51 +02:00
void (*adjust_common_ltp)(struct AACEncContext *s, ChannelElement *cpe);
aacenc: implement the complete AAC-Main profile This commit finalizes AAC-Main profile encoding support by implementing all mandatory and optional tools available in the specifications and current decoders. The AAC-Main profile reqires that prediction support be present (although decoders don't require it to be enabled) for an encoder to be deemed capable of AAC-Main encoding, as well as TNS, PNS and IS, all of which were implemented with previous commits or earlier of this year. Users are encouraged to test the new functionality using either -profile:a aac_main or -aac_pred 1, the former of which will enable the prediction option by default and the latter will change the profile to AAC-Main. No other options shall be changed by enabling either, it's currently up to the users to decide what's best. The current implementation works best using M/S and/or IS, so users are also welcome to enable both options and any other options (TNS, PNS) for maximum quality. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-08-21 20:38:05 +02:00
void (*apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
aacenc_tns: rework coefficient quantization and filter application 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>
2015-09-01 07:44:07 +02:00
void (*apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce);
aacenc: add support for encoding files using Long Term Prediction 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.
2015-10-17 03:22:51 +02:00
void (*update_ltp)(struct AACEncContext *s, SingleChannelElement *sce);
void (*ltp_insert_new_frame)(struct AACEncContext *s);
aacenc: use the new function for setting special band scalefactor indices This commit enables the function added with commit 7c10b87 and uses that new function for setting any special scalefactor indices. This commit does not change the behaviour of the encoder since no bands are being marked as either NOISE_BT(due to the previous PNS implementation removed in the previous commit) or INTENSITY_BT2/INTENSITY_BT. Reviewed-by: Claudio Freire <klaussfreire@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-07-02 20:13:04 +02:00
void (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
aacenc: remove redundant argument from coder functions This commit removes a redundant argument from the functions in aaccoder. The argument lambda was redundant as it was just a copy of s->lambda, to which all functions have access to anyway. This cleans up the function pointers a bit which is helpful as there are a lot of other search_for_* functions under development and with them populated it gets messy. Reviewed-by: Claudio Freire <klaussfreire@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2015-07-29 06:44:24 +02:00
void (*search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
AAC encoder: Extensive improvements 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.
2015-10-11 22:29:50 +02:00
void (*mark_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
aacenc_tns: implement temporal noise shaping 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>
2015-08-21 20:27:38 +02:00
void (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
aacenc: add support for encoding files using Long Term Prediction 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.
2015-10-17 03:22:51 +02:00
void (*search_for_ltp)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
aacenc: remove redundant argument from coder functions This commit removes a redundant argument from the functions in aaccoder. The argument lambda was redundant as it was just a copy of s->lambda, to which all functions have access to anyway. This cleans up the function pointers a bit which is helpful as there are a lot of other search_for_* functions under development and with them populated it gets messy. Reviewed-by: Claudio Freire <klaussfreire@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2015-07-29 06:44:24 +02:00
void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe);
void (*search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
aacenc: implement the complete AAC-Main profile This commit finalizes AAC-Main profile encoding support by implementing all mandatory and optional tools available in the specifications and current decoders. The AAC-Main profile reqires that prediction support be present (although decoders don't require it to be enabled) for an encoder to be deemed capable of AAC-Main encoding, as well as TNS, PNS and IS, all of which were implemented with previous commits or earlier of this year. Users are encouraged to test the new functionality using either -profile:a aac_main or -aac_pred 1, the former of which will enable the prediction option by default and the latter will change the profile to AAC-Main. No other options shall be changed by enabling either, it's currently up to the users to decide what's best. The current implementation works best using M/S and/or IS, so users are also welcome to enable both options and any other options (TNS, PNS) for maximum quality. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-08-21 20:38:05 +02:00
void (*search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce);
Cosmetics: Pretty print the AAC encoder. Originally committed as revision 19376 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:36:45 +03:00
} AACCoefficientsEncoder;
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
extern AACCoefficientsEncoder ff_aac_coders[];
AAC encoder: memoize quantize_band_cost 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.
2015-10-12 08:56:22 +02:00
typedef struct AACQuantizeBandCostCacheEntry {
float rd;
float energy;
aacenc: use generational cache instead of resetting. Approximately 11% faster transcoding from mp3 with default settings. Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
2016-03-06 18:28:42 +02:00
int bits;
AAC encoder: memoize quantize_band_cost 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.
2015-10-12 08:56:22 +02:00
char cb;
char rtz;
aacenc: use generational cache instead of resetting. Approximately 11% faster transcoding from mp3 with default settings. Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
2016-03-06 18:28:42 +02:00
uint16_t generation;
AAC encoder: memoize quantize_band_cost 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.
2015-10-12 08:56:22 +02:00
} AACQuantizeBandCostCacheEntry;
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
/**
* AAC encoder context
*/
typedef struct AACEncContext {
aacenc: Add stereo_mode option. ms_off is the default, until Mid/Side is no longer buggy. Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-06-01 08:38:09 +03:00
AVClass *av_class;
AACEncOptions options; ///< encoding options
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
PutBitContext pb;
Merge FFTContext and MDCTContext Originally committed as revision 19931 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-09-20 20:30:20 +03:00
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
avcodec/aacenc: Use avpriv_float_dsp_alloc() Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-11-29 19:58:13 +02:00
AVFloatDSPContext *fdsp;
aacenc: switch to using the RNG from libavutil 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>
2015-12-14 20:53:09 +02:00
AVLFG lfg; ///< PRNG needed for PNS
aacenc: increase size of s->planar_samples[] from 6 to 8 Left out of last commit which added support for eight channel audio.
2015-10-13 00:25:45 +02:00
float *planar_samples[8]; ///< saved preprocessed input
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
aacenc: do not reject AAC-Main profile This commit permits for the use of the Main profile in encoding. The functionality of that profile will be added in the commits following. By itself, this commit does not alter anything. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-08-21 20:20:22 +02:00
int profile; ///< copied from avctx
aacenc: create and initialize an LTP context This commit only creates and initializes an LTP context which is needed for upcoming commits (TNS). Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-08-21 19:43:09 +02:00
LPCContext lpc; ///< used by TNS
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
int samplerate_index; ///< MPEG-4 samplerate index
aacenc: Store channel count in AACEncContext. Signed-off-by: Alex Converse <alex.converse@gmail.com>
2011-12-08 01:20:10 +03:00
int channels; ///< channel count
aacenc: Make chan_map const correct
2011-06-30 00:33:33 +03:00
const uint8_t *chan_map; ///< channel configuration map
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
ChannelElement *cpe; ///< channel elements
FFPsyContext psy;
struct FFPsyPreprocessContext* psypp;
AACCoefficientsEncoder *coder;
AAC encoder: tweak rate-distortion logic 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).
2015-09-23 07:13:56 +02:00
int cur_channel; ///< current channel for coder context
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
int last_frame;
aacenc: copy PRNG from the decoder Needed for the following PNS commits. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
2015-09-06 16:06:34 +02:00
int random_state;
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
float lambda;
acenc: remove deprecated avctx->frame_bits use 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>
2015-12-18 16:27:13 +02:00
int last_frame_pb_count; ///< number of bits for the previous frame
AAC encoder: Extensive improvements 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.
2015-10-11 22:29:50 +02:00
float lambda_sum; ///< sum(lambda), for Qvg reporting
int lambda_count; ///< count(lambda), for Qvg reporting
AAC encoder: tweak rate-distortion logic 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).
2015-09-23 07:13:56 +02:00
enum RawDataBlockType cur_type; ///< channel group type cur_channel belongs to
aacenc: use AVCodec.encode2()
2012-01-28 20:28:01 +03:00
AudioFrameQueue afq;
aacenc: Use exact values when quantizing, not fuzzy values. This requires us to code small escapes; we can't avoid it. Originally committed as revision 23135 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-05-14 19:49:40 +03:00
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
Add AVX FFT implementation. Signed-off-by: Reinhard Tartler <siretart@tauware.de>
2011-04-25 12:39:01 +03:00
DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients
aacenc: Deinterleave input samples before processing. Signed-off-by: Alex Converse <alex.converse@gmail.com>
2011-12-15 05:43:56 +03:00
aacenc: use generational cache instead of resetting. Approximately 11% faster transcoding from mp3 with default settings. Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
2016-03-06 18:28:42 +02:00
uint16_t quantize_band_cost_cache_generation;
AAC encoder: memoize quantize_band_cost 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.
2015-10-12 08:56:22 +02:00
AACQuantizeBandCostCacheEntry quantize_band_cost_cache[256][128]; ///< memoization area for quantize_band_cost
aacenc: Deinterleave input samples before processing. Signed-off-by: Alex Converse <alex.converse@gmail.com>
2011-12-15 05:43:56 +03:00
struct {
float *samples;
} buffer;
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
} AACEncContext;
mips: Optimization of AAC coefficients encoder functions Signed-off-by: Bojan Zivkovic <bojan@mips.com> Reviewed-by: Nedeljko Babic <Nedeljko.Babic@imgtec.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-03-06 16:55:05 +03:00
void ff_aac_coder_init_mips(AACEncContext *c);
AAC encoder: memoize quantize_band_cost 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.
2015-10-12 08:56:22 +02:00
void ff_quantize_band_cost_cache_init(struct AACEncContext *s);
mips: Optimization of AAC coefficients encoder functions Signed-off-by: Bojan Zivkovic <bojan@mips.com> Reviewed-by: Nedeljko Babic <Nedeljko.Babic@imgtec.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-03-06 16:55:05 +03:00
Merge the AAC encoder from SoC svn. It is still considered experimental. Originally committed as revision 19375 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-07-08 23:01:31 +03:00
#endif /* AVCODEC_AACENC_H */
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