[FFmpeg-devel] [PATCH] AAC decoder round 8

Robert Swain robert.swain
Mon Aug 18 14:47:42 CEST 2008


2008/8/18 Robert Swain <robert.swain at gmail.com>:
> 2008/8/18 Robert Swain <robert.swain at gmail.com>:
>> 2008/8/18 Robert Swain <robert.swain at gmail.com>:
>>> 2008/8/15 Robert Swain <robert.swain at gmail.com>:
>>>> 2008/8/15 Michael Niedermayer <michaelni at gmx.at>:
>>>>> On Fri, Aug 15, 2008 at 09:04:24AM +0100, Robert Swain wrote:
>>>>>> 2008/8/15 Michael Niedermayer <michaelni at gmx.at>:
>>>>>> > On Fri, Aug 15, 2008 at 01:32:08AM +0100, Robert Swain wrote:
>>>>>> >> $subj
>>>>>> >>
>>>>>> >> There's not much left to commit now! :D
>>>>>> >
>>>>>> > ok
>>>>>>
>>>>>> All committed. Just to make it easier for me and/or you to keep track
>>>>>> of, here's another patch attached with the remaining hunks.
>>>>>>
>>>>>> Regards,
>>>>>> Rob
>>>>>
>>>>>> Index: libavcodec/aac.c
>>>>>> ===================================================================
>>>>>> --- libavcodec/aac.c  (revision 14774)
>>>>>> +++ libavcodec/aac.c  (working copy)
>>>>> [...]
>>>>>> @@ -605,6 +616,44 @@
>>>>>>  }
>>>>>>
>>>>>>  /**
>>>>>> + * Decode Temporal Noise Shaping data; reference: table 4.48.
>>>>>> + *
>>>>>> + * @return  Returns error status. 0 - OK, !0 - error
>>>>>> + */
>>>>>> +static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns,
>>>>>> +        GetBitContext * gb, const IndividualChannelStream * ics) {
>>>>>
>>>>>> +    int w, filt, i, coef_len, coef_res = 0, coef_compress;
>>>>>
>>>>> useless init ?
>>>>
>>>> Subtle, but yes.
>>>>
>>>>>> +    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
>>>>>> +    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
>>>>>> +    for (w = 0; w < ics->num_windows; w++) {
>>>>>> +        tns->n_filt[w] = get_bits(gb, 2 - is8);
>>>>>> +
>>>>>> +        if (tns->n_filt[w])
>>>>>> +            coef_res = get_bits1(gb) + 3;
>>>>>> +
>>>>>> +        for (filt = 0; filt < tns->n_filt[w]; filt++) {
>>>>>> +            tns->length[w][filt] = get_bits(gb, 6 - 2*is8);
>>>>>> +
>>>>>
>>>>>> +            if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) <= tns_max_order) {
>>>>>> +                tns->direction[w][filt] = get_bits1(gb);
>>>>>> +                coef_compress = get_bits1(gb);
>>>>>> +                coef_len = coef_res - coef_compress;
>>>>>> +                tns->tmp2_map[w][filt] = tns_tmp2_map[2*coef_compress + coef_res - 3];
>>>>>
>>>>> the 3 can be moved to "coef_len = coef_res - coef_compress + 3"
>>>>
>>>> But, unless I'm missing something, that will change the behaviour of
>>>> the code as more bits will be read in the loop you quoted just below.
>>>>
>>>>>> +
>>>>>> +                for (i = 0; i < tns->order[w][filt]; i++)
>>>>>> +                    tns->coef[w][filt][i] = get_bits(gb, coef_len);
>>>>>
>>>>> tns->coef is only used to index into tmp2_map thus
>>>>> tns->coef could already contain the values from tmp2_map
>>>>> this also would make the tmp2_map field unneeded in the struct
>>>>
>>>> OK, I'll look into this.
>>>>
>>>>>> +            } else {
>>>>>> +                av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.",
>>>>>> +                       tns->order[w][filt], tns_max_order);
>>>>>> +                tns->order[w][filt] = 0;
>>>>>> +                return -1;
>>>>>> +            }
>>>>>
>>>>> if(... > tns_max_order){
>>>>>    ...
>>>>>    return -1
>>>>> }
>>>>> ...
>>>>>
>>>>> seems cleaner to me
>>>>
>>>> Done.
>>>>
>>>>>> +        }
>>>>>> +    }
>>>>>> +    return 0;
>>>>>> +}
>>>>>> +
>>>>>> +/**
>>>>>>   * Decode Mid/Side data; reference: table 4.54.
>>>>>>   *
>>>>>>   * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
>>>>>
>>>>>> @@ -1067,6 +1116,71 @@
>>>>>>  }
>>>>>>
>>>>>>  /**
>>>>>> + * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
>>>>>> + *
>>>>>> + * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
>>>>>> + * @param   coef    spectral coefficients
>>>>>> + */
>>>>>> +static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) {
>>>>>> +    const int mmm = FFMIN(ics->tns_max_bands,  ics->max_sfb);
>>>>>> +    int w, filt, m, i, ib;
>>>>>> +    int bottom, top, order, start, end, size, inc;
>>>>>> +    float tmp;
>>>>>> +    float lpc[TNS_MAX_ORDER + 1], b[2 * TNS_MAX_ORDER];
>>>>>> +
>>>>>> +    for (w = 0; w < ics->num_windows; w++) {
>>>>>> +        bottom = ics->num_swb;
>>>>>> +        for (filt = 0; filt < tns->n_filt[w]; filt++) {
>>>>>> +            top = bottom;
>>>>>> +            bottom = FFMAX(                  0, top - tns->length[w][filt]);
>>>>>
>>>>>> +            order  = FFMIN(tns->order[w][filt], TNS_MAX_ORDER);
>>>>>
>>>>> useless?
>>>>
>>>> Indeed. Removed. Should the 'order' variable be removed as well or is
>>>> accessing it faster than accessing order[][]?
>>>>
>>>>>> +            if (order == 0)
>>>>>> +                continue;
>>>>>> +
>>>>>
>>>>>> +            // tns_decode_coef
>>>>>> +            lpc[0] = 1;
>>>>>> +            for (m = 1; m <= order; m++) {
>>>>>> +                lpc[m] = tns->tmp2_map[w][filt][tns->coef[w][filt][m - 1]];
>>>>>> +                for (i = 1; i < m; i++)
>>>>>> +                    b[i] = lpc[i] + lpc[m] * lpc[m-i];
>>>>>> +                for (i = 1; i < m; i++)
>>>>>> +                    lpc[i] = b[i];
>>>>>> +            }
>>>>>
>>>>> looks a little like eval_coefs from ra144.c
>>>>> but later is fixedpoint, so this is more a random comment than anything
>>>>>
>>>>>
>>>>>> +
>>>>>> +            start = ics->swb_offset[FFMIN(bottom, mmm)];
>>>>>> +            end   = ics->swb_offset[FFMIN(   top, mmm)];
>>>>>> +            if ((size = end - start) <= 0)
>>>>>> +                continue;
>>>>>> +            if (tns->direction[w][filt]) {
>>>>>> +                inc = -1; start = end - 1;
>>>>>> +            } else {
>>>>>> +                inc = 1;
>>>>>> +            }
>>>>>> +            start += w * 128;
>>>>>> +
>>>>>> +            // ar filter
>>>>>> +            memset(b, 0, sizeof(b));
>>>>>> +            ib = 0;
>>>>>
>>>>>> +            for (m = 0; m < size; m++) {
>>>>>> +                tmp = coef[start];
>>>>>> +                if (decode) {
>>>>>> +                    for (i = 0; i < order; i++)
>>>>>> +                        tmp -= b[ib + i] * lpc[i + 1];
>>>>>> +                } else { // encode
>>>>>> +                    for (i = 0; i < order; i++)
>>>>>> +                        tmp += b[i]      * lpc[i + 1];
>>>>>> +                }
>>>>>> +                if (--ib < 0)
>>>>>> +                    ib = order - 1;
>>>>>> +                b[ib] = b[ib + order] = tmp;
>>>>>> +                coef[start] = tmp;
>>>>>> +                start += inc;
>>>>>> +            }
>>>>>
>>>>> decode is always 1
>>>>
>>>> This is left from LTP. I'll remove it when submitting next.
>>>>
>>>>> b is not truly needed, coef[] can be used i its place
>>>>> also this is likely relevant to overal codec speed so it
>>>>> should be written more with speed than compactness in mind
>>>>
>>>> Does that mean you want me to rewrite it?
>>>
>>> Without the encode case it would look like this, I think:
>>>
>>> memset(b, 0, sizeof(b));
>>> ib = 0;
>>> for (m = 0; m < size; m++) {
>>>    tmp = coef[start];
>>>    for (i = 0; i < order; i++)
>>>        tmp -= b[i] * lpc[i + 1];
>>>    if (--ib < 0)
>>>        ib = order - 1;
>>>    coef[start] = b[ib] = tmp;
>>>    start += inc;
>>> }
>>>
>>> Then I don't think tmp is needed or b as none of the values of coeff
>>> but the one currently being filtered are changed and that one is only
>>> subtracted from, it's not involved in any calculations, so I think the
>>> following is correct:
>>>
>>> for (m = 0; m < size; m++) {
>>>    for (i = 0; i < FFMIN(m, order); i++)
>>>        coef[start] -= coef[start - 1 - i] * lpc[i + 1];
>>>    start += inc;
>>> }
>>
>> Make that:
>>
>> for (m = 0; m < size; m++) {
>>   for (i = 1; i <= FFMIN(m, order); i++)
>>       coef[start] -= coef[start - i] * lpc[i];
>>   start += inc;
>> }
>
> Or, if you like, the start incrementation can be merged into the outer
> for() loop's incrementation stuff. What is that bit technically
> called? The m++ section.

See attached. Tested with the zodiac clip which is known to use TNS
and produces identical PSNR, so I'm assuming identical output with no
known regressions.

Rob
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