ffmpeg/libavcodec/aacenc.h

142 lines
5.7 KiB
C
Raw Normal View History

/*
* 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
#include "libavutil/float_dsp.h"
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "put_bits.h"
#include "aac.h"
2012-01-28 17:28:01 +00:00
#include "audio_frame_queue.h"
#include "psymodel.h"
#include "lpc.h"
typedef enum AACCoder {
AAC_CODER_ANMR = 0,
AAC_CODER_TWOLOOP,
AAC_CODER_FAST,
AAC_CODER_NB,
}AACCoder;
typedef struct AACEncOptions {
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 11:18:42 +00:00
int pns;
int tns;
int ltp;
int pred;
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 18:13:07 +00:00
int intensity_stereo;
} AACEncOptions;
struct AACEncContext;
typedef struct AACCoefficientsEncoder {
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);
void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size,
int scale_idx, int cb, const float lambda, int rtz);
void (*encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce);
void (*encode_ltp_info)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
void (*encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
void (*adjust_common_pred)(struct AACEncContext *s, ChannelElement *cpe);
void (*adjust_common_ltp)(struct AACEncContext *s, ChannelElement *cpe);
void (*apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
void (*apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce);
void (*update_ltp)(struct AACEncContext *s, SingleChannelElement *sce);
void (*ltp_insert_new_frame)(struct AACEncContext *s);
void (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
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 20:29:50 +00:00
void (*mark_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
void (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
void (*search_for_ltp)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe);
void (*search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
void (*search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce);
} AACCoefficientsEncoder;
extern AACCoefficientsEncoder ff_aac_coders[];
typedef struct AACQuantizeBandCostCacheEntry {
float rd;
float energy;
int bits;
char cb;
char rtz;
uint16_t generation;
} AACQuantizeBandCostCacheEntry;
/**
* AAC encoder context
*/
typedef struct AACEncContext {
AVClass *av_class;
AACEncOptions options; ///< encoding options
PutBitContext pb;
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
AVFloatDSPContext *fdsp;
AVLFG lfg; ///< PRNG needed for PNS
float *planar_samples[8]; ///< saved preprocessed input
int profile; ///< copied from avctx
LPCContext lpc; ///< used by TNS
int samplerate_index; ///< MPEG-4 samplerate index
int channels; ///< channel count
2011-06-29 21:33:33 +00:00
const uint8_t *chan_map; ///< channel configuration map
ChannelElement *cpe; ///< channel elements
FFPsyContext psy;
struct FFPsyPreprocessContext* psypp;
AACCoefficientsEncoder *coder;
int cur_channel; ///< current channel for coder context
int last_frame;
int random_state;
float lambda;
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 20:29:50 +00:00
float lambda_sum; ///< sum(lambda), for Qvg reporting
int lambda_count; ///< count(lambda), for Qvg reporting
enum RawDataBlockType cur_type; ///< channel group type cur_channel belongs to
2012-01-28 17:28:01 +00:00
AudioFrameQueue afq;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients
uint16_t quantize_band_cost_cache_generation;
AACQuantizeBandCostCacheEntry quantize_band_cost_cache[256][128]; ///< memoization area for quantize_band_cost
struct {
float *samples;
} buffer;
} AACEncContext;
void ff_aac_coder_init_mips(AACEncContext *c);
void ff_quantize_band_cost_cache_init(struct AACEncContext *s);
#endif /* AVCODEC_AACENC_H */