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

Robert Swain robert.swain
Mon Aug 18 14:31:55 CEST 2008


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.

Rob




More information about the ffmpeg-devel mailing list