[FFmpeg-devel] [PATCH] Implement AAC Long Term Prediction (LTP) decoding module
Young Han Lee
cpumaker
Thu Feb 3 12:37:19 CET 2011
On Thu, Feb 3, 2011 at 2:18 PM, Alex Converse <alex.converse at gmail.com>wrote:
> On Mon, Jan 31, 2011 at 5:57 PM, Young Han Lee <cpumaker at gmail.com> wrote:
>
> [...]
>
> > I'm little late because of business trip.
> >
> > please check it again. :)
> >
> > Young Han
> >
>
> Great start so far.
>
> > diff --git a/libavcodec/aac.h b/libavcodec/aac.h
> > index 714e314..0a94785 100644
> > --- a/libavcodec/aac.h
> > +++ b/libavcodec/aac.h
> > @@ -42,6 +42,7 @@
> > #define MAX_ELEM_ID 16
> >
> > #define TNS_MAX_ORDER 20
> > +#define MAX_LTP_LONG_SFB 40
> >
> > enum RawDataBlockType {
> > TYPE_SCE,
> > @@ -129,6 +130,18 @@ typedef struct {
> > #define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference
> allowed by standard
> > #define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero
> scalefactor indices difference
> >
> > +
> > +/**
> > + * Long Term Prediction
> > + */
> > +typedef struct {
> > + int present;
> > + int lag;
> > + float coef;
> > + int used[MAX_LTP_LONG_SFB];
>
> This can be an int8_t
>
> It is changed and the others are also changed.
> > +} LongTermPrediction;
> > +
> > +
> > /**
> > * Individual Channel Stream
> > */
> > @@ -138,6 +151,8 @@ typedef struct {
> > uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window,
> otherwise use a sinus window.
> > int num_window_groups;
> > uint8_t group_len[8];
> > + LongTermPrediction ltp;
> > + LongTermPrediction ltp2;
>
> Let's keep only one set of LTP variables per ICS, see my notes below for
> details
>
>
see the note below for details
> > const uint16_t *swb_offset; ///< table of offsets to the lowest
> spectral coefficient of a scalefactor band, sfb, for a particular window
> > const uint8_t *swb_sizes; ///< table of scalefactor band sizes for
> a particular window
> > int num_swb; ///< number of scalefactor window bands
> > @@ -212,6 +227,8 @@ typedef struct {
> > uint8_t zeroes[128]; ///< band is not coded
> (used by encoder)
> > DECLARE_ALIGNED(16, float, coeffs)[1024]; ///< coefficients for
> IMDCT
> > DECLARE_ALIGNED(16, float, saved)[1024]; ///< overlap
> > + DECLARE_ALIGNED(16, float, saved_ltp)[1024]; ///< overlap for LTP
> > + int16_t ltp_state[3072];
>
> This needs to be aligned
>
You mean the alignment of comment? I just applied.
> > DECLARE_ALIGNED(16, float, ret)[2048]; ///< PCM output
> > PredictorState predictor_state[MAX_PREDICTORS];
> > } SingleChannelElement;
> > @@ -258,7 +275,7 @@ typedef struct {
> > * @defgroup temporary aligned temporary buffers (We do not want to
> have these on the stack.)
> > * @{
> > */
> > - DECLARE_ALIGNED(16, float, buf_mdct)[1024];
> > + DECLARE_ALIGNED(16, float, buf_mdct)[2048];
> > /** @} */
> >
> > /**
> > @@ -267,6 +284,7 @@ typedef struct {
> > */
> > FFTContext mdct;
> > FFTContext mdct_small;
> > + FFTContext mdct_ltp;
> > DSPContext dsp;
> > int random_state;
> > /** @} */
> > diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
> > index 2127099..249498c 100644
> > --- a/libavcodec/aacdec.c
> > +++ b/libavcodec/aacdec.c
> > @@ -477,6 +477,7 @@ static int decode_audio_specific_config(AACContext
> *ac,
> > switch (m4ac->object_type) {
> > case AOT_AAC_MAIN:
> > case AOT_AAC_LC:
> > + case AOT_AAC_LTP:
> > if (decode_ga_specific_config(ac, avctx, &gb, m4ac,
> m4ac->chan_config))
> > return -1;
> > break;
> > @@ -578,8 +579,9 @@ static av_cold int aac_decode_init(AVCodecContext
> *avctx)
> > ff_aac_scalefactor_code,
> sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
> > 352);
> >
> > - ff_mdct_init(&ac->mdct, 11, 1, 1.0);
> > - ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
> > + ff_mdct_init(&ac->mdct, 11, 1, 1.0);
> > + ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
> > + ff_mdct_init(&ac->mdct_ltp, 11, 0, 1.0);
> > // window initialization
> > ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
> > ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
> > @@ -629,6 +631,24 @@ static int decode_prediction(AACContext *ac,
> IndividualChannelStream *ics,
> > }
> >
> > /**
> > + * Decode Long Term Prediction data; reference: table 4.xx.
> > + */
> > +static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
> > + GetBitContext *gb, uint8_t max_sfb)
> > +{
> > + int sfb;
> > + if (ac->m4ac.object_type == AOT_ER_AAC_LD) {
> > + av_log(ac->avctx, AV_LOG_ERROR, "LTP is not supported in ER AAC
> LD .\n");
>
> No ER syntax is currently supported so this check is unnecessary.
>
The message is erased.
> > + } else {
> > + ltp->lag = get_bits(gb, 11);
> > + ltp->coef = ltp_coef[get_bits(gb, 3)] * ac->sf_scale;
> > + for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
> > + ltp->used[sfb] = get_bits1(gb);
> > + }
> > +}
> > +
> > +
> > +/**
> > * Decode Individual Channel Stream info; reference: table 4.6.
> > *
> > * @param common_window Channels have independent [0], or shared
> [1], Individual Channel Stream information.
> > @@ -682,9 +702,11 @@ static int decode_ics_info(AACContext *ac,
> IndividualChannelStream *ics,
> > memset(ics, 0, sizeof(IndividualChannelStream));
> > return -1;
> > } else {
> > - av_log_missing_feature(ac->avctx, "Predictor bit set but
> LTP is", 1);
> > - memset(ics, 0, sizeof(IndividualChannelStream));
> > - return -1;
> > + if ((ics->ltp.present = get_bits(gb, 1)))
> > + decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
> > + if (common_window)
> > + if ((ics->ltp2.present = get_bits(gb, 1)))
> > + decode_ltp(ac, &ics->ltp2, gb, ics->max_sfb);
>
> Let's check for this second set of LTP parameters when we return from
> ice_info in the common_window case
>
>
the ltp2 can be eliminated by moving the second "decode_ltp" function to
outside of ice_info.
Thanks for good comment:)
> > }
> > }
> > }
> > @@ -1418,6 +1440,7 @@ static int decode_cpe(AACContext *ac, GetBitContext
> *gb, ChannelElement *cpe)
> > i = cpe->ch[1].ics.use_kb_window[0];
> > cpe->ch[1].ics = cpe->ch[0].ics;
> > cpe->ch[1].ics.use_kb_window[1] = i;
> > + cpe->ch[1].ics.ltp = cpe->ch[0].ics.ltp2;
> > ms_present = get_bits(gb, 2);
> > if (ms_present == 3) {
> > av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is
> reserved.\n");
> > @@ -1657,6 +1680,7 @@ static void apply_tns(float coef[1024],
> TemporalNoiseShaping *tns,
> > int w, filt, m, i;
> > int bottom, top, order, start, end, size, inc;
> > float lpc[TNS_MAX_ORDER];
> > + float tmp[1024];
>
> This is a fairly large stack allocation. If it can't be shrunk let's
> put it in the context or reuse one of the other scratch buffers in the
> context.
>
>
To reduce the stack, ma filtering is slightly changed. (a loop is added)
> >
> > for (w = 0; w < ics->num_windows; w++) {
> > bottom = ics->num_swb;
> > @@ -1682,23 +1706,108 @@ static void apply_tns(float coef[1024],
> TemporalNoiseShaping *tns,
> > }
> > start += w * 128;
> >
> > - // ar filter
> > - for (m = 0; m < size; m++, start += inc)
> > - for (i = 1; i <= FFMIN(m, order); i++)
> > - coef[start] -= coef[start - i * inc] * lpc[i - 1];
> > + if (decode) {
> > + // ar filter
> > + for (m = 0; m < size; m++, start += inc)
> > + for (i = 1; i <= FFMIN(m, order); i++)
> > + coef[start] -= coef[start - i * inc] * lpc[i -
> 1];
> > + } else {
> > + // ma filter
> > + for (m = 0; m < size; m++, start += inc) {
> > + tmp[start] = coef[start];
> > + for (i = 1; i <= FFMIN(m, order); i++)
> > + coef[start] += tmp[start - i * inc] * lpc[i -
> 1];
> > + }
> > + }
> > }
> > }
> > }
> >
> > +
> > +
> > +/**
> > + * Windowing and MDCT to obtain the spectral coefficient from the
> predicted sample by LTP
> > + */
> > +static void windowing_and_mdct_ltp(AACContext *ac, float *out,
> > + float *in, IndividualChannelStream
> *ics)
> > +{
> > + const float * lwindow = ics->use_kb_window[0] ?
> ff_aac_kbd_long_1024 : ff_sine_1024;
> > + const float * swindow = ics->use_kb_window[0] ?
> ff_aac_kbd_short_128 : ff_sine_128;
> > + const float * lwindow_prev = ics->use_kb_window[1] ?
> ff_aac_kbd_long_1024 : ff_sine_1024;
> > + const float * swindow_prev = ics->use_kb_window[1] ?
> ff_aac_kbd_short_128 : ff_sine_128;
> > + float * buf = ac->buf_mdct;
> > +
> > + if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
> > + ac->dsp.vector_fmul(buf, in, lwindow_prev, 1024);
> > + } else {
> > + memset(buf, 0, 448 * sizeof(float));
> > + ac->dsp.vector_fmul(buf + 448, in + 448, swindow_prev, 128);
> > + memcpy(buf + 576, in + 576, 448 * sizeof(float));
> > + }
> > + if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
> > + ac->dsp.vector_fmul_reverse(buf + 1024, in + 1024, lwindow,
> 1024);
> > + } else {
> > + memcpy(buf + 1024, in + 1024, 448 * sizeof(float));
> > + ac->dsp.vector_fmul_reverse(buf + 1024 + 448, in + 1024 + 448,
> swindow, 128);
> > + memset(buf + 1024 + 576, 0, 448 * sizeof(float));
> > + }
> > + ff_mdct_calc(&ac->mdct_ltp, out, buf);
> > +}
> > +
> > +
> > +/**
> > + * Apply the long term prediction
> > + */
> > +static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
> > +{
> > + const LongTermPrediction *ltp = &sce->ics.ltp;
> > + const uint16_t *offsets = sce->ics.swb_offset;
> > + int i, sfb;
> > +
> > + if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
> > + float x_est[2048], X_est[1024];
> > + int16_t num_samples = 2048;
> > + if (ltp->lag < 1024)
> > + num_samples = ltp->lag + 1024;
> > + for (i = 0; i < num_samples; i++)
> > + x_est[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
> > + for ( ; i < 2048; i++)
> > + x_est[i] = 0.0f;
> > +
> > + windowing_and_mdct_ltp(ac, X_est, x_est, &sce->ics);
> > +
> > + if (sce->tns.present)
> > + apply_tns(X_est, &sce->tns, &sce->ics, 0);
> > +
> > + for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB);
> sfb++)
> > + if (ltp->used[sfb])
> > + for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
> > + sce->coeffs[i] += X_est[i];
> > + }
> > +}
> > +
> > +/**
> > + * Update the LTP buffer for next frame
> > + */
> > +static void update_ltp(AACContext *ac, SingleChannelElement *sce)
> > +{
> > + memcpy(sce->ltp_state, &sce->ltp_state[1024], 1024 *
> sizeof(int16_t));
> > + ac->dsp.float_to_int16(&(sce->ltp_state[1024]), sce->ret,
> 1024);
> > + ac->dsp.float_to_int16(&(sce->ltp_state[2048]), sce->saved_ltp,
> 1024);
> > +}
> > +
> > +
> > /**
> > * Conduct IMDCT and windowing.
> > */
> > static void imdct_and_windowing(AACContext *ac, SingleChannelElement
> *sce)
> > {
> > IndividualChannelStream *ics = &sce->ics;
> > - float *in = sce->coeffs;
> > - float *out = sce->ret;
> > - float *saved = sce->saved;
> > + float *in = sce->coeffs;
> > + float *out = sce->ret;
> > + float *saved = sce->saved;
> > + float *saved_ltp = sce->saved_ltp;
> > + const float *lwindow = ics->use_kb_window[0] ?
> ff_aac_kbd_long_1024 : ff_sine_1024;
> > const float *swindow = ics->use_kb_window[0] ?
> ff_aac_kbd_short_128 : ff_sine_128;
> > const float *lwindow_prev = ics->use_kb_window[1] ?
> ff_aac_kbd_long_1024 : ff_sine_1024;
> > const float *swindow_prev = ics->use_kb_window[1] ?
> ff_aac_kbd_short_128 : ff_sine_128;
> > @@ -1713,6 +1822,9 @@ static void imdct_and_windowing(AACContext *ac,
> SingleChannelElement *sce)
> > } else
> > ff_imdct_half(&ac->mdct, buf, in);
> >
> > + for (i = 0; i < 512; i++)
> > + buf[1535 - i] = buf[512 + i];
> > +
>
> While it's good to have LTP for completeness the typical use case of
> the decoder will still be non-LTP streams.
>
> As a result we should try not to inconvenience the decoder with extra
> computation in the non-LTP case.
>
> Put this extra copying behind a guard to make sure we are only doing
> it on LTP streams.
>
> The same applies for the other code added in this function
>
>
The extra computation is for the next frame.
In order to do it only on the LTP streams,
the codes are moved to "update_ltp()" before the update,
and the location of "update_ltp()" is moved to
right after "imdct_and_windowing()"
> > /* window overlapping
> > * NOTE: To simplify the overlapping code, all 'meaningless' short
> to long
> > * and long to short transitions are considered to be short to short
> > @@ -1745,11 +1857,18 @@ static void imdct_and_windowing(AACContext *ac,
> SingleChannelElement *sce)
> > ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf +
> 6*128, swindow, 0, 64);
> > ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf +
> 7*128, swindow, 0, 64);
> > memcpy( saved + 448, buf + 7*128 + 64, 64 *
> sizeof(float));
> > + memcpy( saved_ltp, saved, 512 *
> sizeof(float));
> > + ac->dsp.vector_fmul_reverse(saved_ltp+448, buf + 960,
> swindow, 128);
> > + memset( saved_ltp + 576, 0, 448 *
> sizeof(float));
> > } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
> > memcpy( saved, buf + 512, 448 *
> sizeof(float));
> > memcpy( saved + 448, buf + 7*128 + 64, 64 *
> sizeof(float));
> > + memcpy( saved_ltp, buf + 512, 448 *
> sizeof(float));
> > + ac->dsp.vector_fmul_reverse(saved_ltp+448, buf + 960,
> swindow, 128);
> > + memset( saved_ltp + 576, 0, 448 *
> sizeof(float));
> > } else { // LONG_STOP or ONLY_LONG
> > memcpy( saved, buf + 512, 512 *
> sizeof(float));
> > + ac->dsp.vector_fmul_reverse(saved_ltp, buf + 512,
> lwindow, 1024);
> > }
> > }
> >
> > @@ -1855,6 +1974,14 @@ static void spectral_to_sample(AACContext *ac)
> > if (che) {
> > if (type <= TYPE_CPE)
> > apply_channel_coupling(ac, che, type, i, BEFORE_TNS,
> apply_dependent_coupling);
> > + if (che->ch[0].ics.predictor_present) {
> > + if (ac->m4ac.object_type == AOT_AAC_LTP) {
> > + if (che->ch[0].ics.ltp.present)
> > + apply_ltp(ac, &che->ch[0]);
> > + if (che->ch[1].ics.ltp.present && type ==
> TYPE_CPE)
> > + apply_ltp(ac, &che->ch[1]);
> > + }
> > + }
> > if (che->ch[0].tns.present)
> > apply_tns(che->ch[0].coeffs, &che->ch[0].tns,
> &che->ch[0].ics, 1);
> > if (che->ch[1].tns.present)
> > @@ -1872,6 +1999,11 @@ static void spectral_to_sample(AACContext *ac)
> > }
> > if (type <= TYPE_CCE)
> > apply_channel_coupling(ac, che, type, i,
> AFTER_IMDCT, apply_independent_coupling);
> > + if (ac->m4ac.object_type == AOT_AAC_LTP) {
> > + update_ltp(ac, &che->ch[0]);
> > + if (type == TYPE_CPE)
> > + update_ltp(ac, &che->ch[1]);
> > + }
> > }
> > }
> > }
> > @@ -2078,6 +2210,7 @@ static av_cold int aac_decode_close(AVCodecContext
> *avctx)
> >
> > ff_mdct_end(&ac->mdct);
> > ff_mdct_end(&ac->mdct_small);
> > + ff_mdct_end(&ac->mdct_ltp);
> > return 0;
> > }
> >
> > diff --git a/libavcodec/aacdectab.h b/libavcodec/aacdectab.h
> > index b4307f1..500e8f2 100644
> > --- a/libavcodec/aacdectab.h
> > +++ b/libavcodec/aacdectab.h
> OK
>
> [...]
>
> Regards,
> Alex Converse
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel at mplayerhq.hu
> https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel
>
please check again.
Young Han
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