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authorClaudio Freire <klaussfreire@gmail.com>2015-10-11 17:29:50 -0300
committerClaudio Freire <klaussfreire@gmail.com>2015-10-11 17:29:50 -0300
commit01ecb7172b684f1c4b3e748f95c5a9a494ca36ec (patch)
tree5f724b1e5ea315dfeab49a97d15cac150d29437c /libavcodec/psymodel.c
parent624057df3fd5b0044eeed94d2b8e14105b8944dc (diff)
downloadffmpeg-01ecb7172b684f1c4b3e748f95c5a9a494ca36ec.tar.gz
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.
Diffstat (limited to 'libavcodec/psymodel.c')
-rw-r--r--libavcodec/psymodel.c12
1 files changed, 4 insertions, 8 deletions
diff --git a/libavcodec/psymodel.c b/libavcodec/psymodel.c
index 824eefb79e..f7bca6890c 100644
--- a/libavcodec/psymodel.c
+++ b/libavcodec/psymodel.c
@@ -109,25 +109,21 @@ av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *av
return NULL;
ctx->avctx = avctx;
+ /* AAC has its own LP method */
+ if (avctx->codec_id != AV_CODEC_ID_AAC) {
if (avctx->cutoff > 0)
cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
- if (!cutoff_coeff && avctx->codec_id == AV_CODEC_ID_AAC)
- cutoff_coeff = 2.0 * AAC_CUTOFF(avctx) / avctx->sample_rate;
-
if (cutoff_coeff && cutoff_coeff < 0.98)
ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
FF_FILTER_MODE_LOWPASS, FILT_ORDER,
cutoff_coeff, 0.0, 0.0);
if (ctx->fcoeffs) {
- ctx->fstate = av_mallocz_array(sizeof(ctx->fstate[0]), avctx->channels);
- if (!ctx->fstate) {
- av_free(ctx);
- return NULL;
- }
+ ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
for (i = 0; i < avctx->channels; i++)
ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
}
+ }
ff_iir_filter_init(&ctx->fiir);