Chiebot-Mirror
/
FFmpeg
8
0
Fork 0
Code Issues Pull Requests Projects Releases Wiki Activity

More OKed sections of AAC decoder code

Browse Source 
Originally committed as revision 14770 to svn://svn.ffmpeg.org/ffmpeg/trunk
pull/126/head
Robert Swain 17 years ago
parent e9b9a2f86f
commit 62a57fae59
2 changed files with 347 additions and 0 deletions
Whitespace
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. 301
      libavcodec/aac.c
  2. 46
      libavcodec/aac.h

301
libavcodec/aac.c
Unescape View File

@ -99,6 +99,53 @@ static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
/**
* Configure output channel order based on the current program configuration element.
*
* @param che_pos current channel position configuration
* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID],
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]) {
AVCodecContext *avctx = ac->avccontext;
int i, type, channels = 0;
if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])))
return 0; /* no change */
memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
/* Allocate or free elements depending on if they are in the
* current program configuration.
*
* Set up default 1:1 output mapping.
*
* For a 5.1 stream the output order will be:
* [ Front Left ] [ Front Right ] [ Center ] [ LFE ] [ Surround Left ] [ Surround Right ]
*/
for(i = 0; i < MAX_ELEM_ID; i++) {
for(type = 0; type < 4; type++) {
if(che_pos[type][i]) {
if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement))))
return AVERROR(ENOMEM);
if(type != TYPE_CCE) {
ac->output_data[channels++] = ac->che[type][i]->ch[0].ret;
if(type == TYPE_CPE) {
ac->output_data[channels++] = ac->che[type][i]->ch[1].ret;
}
}
} else
av_freep(&ac->che[type][i]);
}
}
avctx->channels = channels;
return 0;
}
/**
* Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
*
@ -209,6 +256,17 @@ static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_c
return 0;
}
/**
* Decode GA "General Audio" specific configuration; reference: table 4.1.
*
* @return Returns error status. 0 - OK, !0 - error
*/
static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) {
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
int extension_flag, ret;
if(get_bits1(gb)) { // frameLengthFlag
av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
return -1;
}
@ -289,6 +347,17 @@ static int decode_audio_specific_config(AACContext * ac, void *data, int data_si
return 0;
}
/**
* linear congruential pseudorandom number generator
*
* @param previous_val pointer to the current state of the generator
*
* @return Returns a 32-bit pseudorandom integer
*/
static av_always_inline int lcg_random(int previous_val) {
return previous_val * 1664525 + 1013904223;
}
static av_cold int aac_decode_init(AVCodecContext * avccontext) {
AACContext * ac = avccontext->priv_data;
int i;
@ -381,6 +450,21 @@ static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBi
ics->num_window_groups = 1;
ics->group_len[0] = 1;
if (get_bits1(gb)) {
av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
}
}
if(ics->max_sfb > ics->num_swb) {
av_log(ac->avccontext, AV_LOG_ERROR,
"Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
ics->max_sfb, ics->num_swb);
memset(ics, 0, sizeof(IndividualChannelStream));
return -1;
}
return 0;
}
@ -520,6 +604,14 @@ static void decode_pulses(Pulse * pulse, GetBitContext * gb) {
*/
static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
int ms_present) {
int idx;
if (ms_present == 1) {
for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
cpe->ms_mask[idx] = get_bits1(gb);
} else if (ms_present == 2) {
memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
}
}
/**
* Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3.
@ -635,6 +727,64 @@ static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
return 0;
}
coup->coupling_point = 2*get_bits1(gb);
coup->num_coupled = get_bits(gb, 3);
for (c = 0; c <= coup->num_coupled; c++) {
num_gain++;
coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
coup->id_select[c] = get_bits(gb, 4);
if (coup->type[c] == TYPE_CPE) {
coup->ch_select[c] = get_bits(gb, 2);
if (coup->ch_select[c] == 3)
num_gain++;
} else
coup->ch_select[c] = 1;
}
coup->coupling_point += get_bits1(gb);
if (coup->coupling_point == 2) {
av_log(ac->avccontext, AV_LOG_ERROR,
"Independently switched CCE with 'invalid' domain signalled.\n");
memset(coup, 0, sizeof(ChannelCoupling));
return -1;
}
sign = get_bits(gb, 1);
scale = pow(2., pow(2., get_bits(gb, 2) - 3));
if ((ret = decode_ics(ac, sce, gb, 0, 0)))
return ret;
for (c = 0; c < num_gain; c++) {
int cge = 1;
int gain = 0;
float gain_cache = 1.;
if (c) {
cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
gain_cache = pow(scale, gain);
}
for (g = 0; g < sce->ics.num_window_groups; g++)
for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++)
if (sce->band_type[idx] != ZERO_BT) {
if (!cge) {
int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (t) {
int s = 1;
if (sign) {
s -= 2 * (t & 0x1);
t >>= 1;
}
gain += t;
gain_cache = pow(scale, gain) * s;
}
}
coup->gain[c][idx] = gain_cache;
}
}
return 0;
}
/**
* Decode Spectral Band Replication extension data; reference: table 4.55.
*
@ -650,6 +800,23 @@ static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, in
return cnt;
}
/**
* Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
*
* @return Returns number of bytes consumed.
*/
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) {
int i;
int num_excl_chan = 0;
do {
for (i = 0; i < 7; i++)
che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
} while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
return num_excl_chan / 7;
}
/**
* Decode dynamic range information; reference: table 4.52.
*
@ -746,6 +913,55 @@ static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {
float * buf = ac->buf_mdct;
int i;
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
av_log(ac->avccontext, AV_LOG_WARNING,
"Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
"If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
for (i = 0; i < 2048; i += 256) {
ff_imdct_calc(&ac->mdct_small, buf + i, in + i/2);
ac->dsp.vector_fmul_reverse(ac->revers + i/2, buf + i + 128, swindow, 128);
}
for (i = 0; i < 448; i++) out[i] = saved[i] + ac->add_bias;
ac->dsp.vector_fmul_add_add(out + 448 + 0*128, buf + 0*128, swindow_prev, saved + 448 , ac->add_bias, 128, 1);
ac->dsp.vector_fmul_add_add(out + 448 + 1*128, buf + 2*128, swindow, ac->revers + 0*128, ac->add_bias, 128, 1);
ac->dsp.vector_fmul_add_add(out + 448 + 2*128, buf + 4*128, swindow, ac->revers + 1*128, ac->add_bias, 128, 1);
ac->dsp.vector_fmul_add_add(out + 448 + 3*128, buf + 6*128, swindow, ac->revers + 2*128, ac->add_bias, 128, 1);
ac->dsp.vector_fmul_add_add(out + 448 + 4*128, buf + 8*128, swindow, ac->revers + 3*128, ac->add_bias, 64, 1);
#if 0
vector_fmul_add_add_add(&ac->dsp, out + 448 + 1*128, buf + 2*128, swindow, saved + 448 + 1*128, ac->revers + 0*128, ac->add_bias, 128);
vector_fmul_add_add_add(&ac->dsp, out + 448 + 2*128, buf + 4*128, swindow, saved + 448 + 2*128, ac->revers + 1*128, ac->add_bias, 128);
vector_fmul_add_add_add(&ac->dsp, out + 448 + 3*128, buf + 6*128, swindow, saved + 448 + 3*128, ac->revers + 2*128, ac->add_bias, 128);
vector_fmul_add_add_add(&ac->dsp, out + 448 + 4*128, buf + 8*128, swindow, saved + 448 + 4*128, ac->revers + 3*128, ac->add_bias, 64);
#endif
ac->dsp.vector_fmul_add_add(saved, buf + 1024 + 64, swindow + 64, ac->revers + 3*128+64, 0, 64, 1);
ac->dsp.vector_fmul_add_add(saved + 64, buf + 1024 + 2*128, swindow, ac->revers + 4*128, 0, 128, 1);
ac->dsp.vector_fmul_add_add(saved + 192, buf + 1024 + 4*128, swindow, ac->revers + 5*128, 0, 128, 1);
ac->dsp.vector_fmul_add_add(saved + 320, buf + 1024 + 6*128, swindow, ac->revers + 6*128, 0, 128, 1);
memcpy( saved + 448, ac->revers + 7*128, 128 * sizeof(float));
memset( saved + 576, 0, 448 * sizeof(float));
} else {
ff_imdct_calc(&ac->mdct, buf, in);
if (ics->window_sequence[0] == LONG_STOP_SEQUENCE) {
for (i = 0; i < 448; i++) out[i] = saved[i] + ac->add_bias;
ac->dsp.vector_fmul_add_add(out + 448, buf + 448, swindow_prev, saved + 448, ac->add_bias, 128, 1);
for (i = 576; i < 1024; i++) out[i] = buf[i] + saved[i] + ac->add_bias;
} else {
ac->dsp.vector_fmul_add_add(out, buf, lwindow_prev, saved, ac->add_bias, 1024, 1);
}
if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
memcpy(saved, buf + 1024, 448 * sizeof(float));
ac->dsp.vector_fmul_reverse(saved + 448, buf + 1024 + 448, swindow, 128);
memset(saved + 576, 0, 448 * sizeof(float));
} else {
ac->dsp.vector_fmul_reverse(saved, buf + 1024, lwindow, 1024);
}
}
}
/**
* Apply dependent channel coupling (applied before IMDCT).
*
@ -789,6 +1005,91 @@ static void apply_independent_coupling(AACContext * ac, SingleChannelElement * s
sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias);
}
}
}
}
}
static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) {
AACContext * ac = avccontext->priv_data;
GetBitContext gb;
enum RawDataBlockType elem_type;
int err, elem_id, data_size_tmp;
init_get_bits(&gb, buf, buf_size*8);
// parse
while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
elem_id = get_bits(&gb, 4);
err = -1;
if(elem_type == TYPE_SCE && elem_id == 1 &&
!ac->che[TYPE_SCE][elem_id] && ac->che[TYPE_LFE][0]) {
/* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have
encountered such a stream, transfer the LFE[0] element to SCE[1] */
ac->che[TYPE_SCE][elem_id] = ac->che[TYPE_LFE][0];
ac->che[TYPE_LFE][0] = NULL;
}
if(elem_type && elem_type < TYPE_DSE) {
if(!ac->che[elem_type][elem_id])
return -1;
if(elem_type != TYPE_CCE)
ac->che[elem_type][elem_id]->coup.coupling_point = 4;
}
switch (elem_type) {
case TYPE_SCE:
err = decode_ics(ac, &ac->che[TYPE_SCE][elem_id]->ch[0], &gb, 0, 0);
break;
case TYPE_CPE:
err = decode_cpe(ac, &gb, elem_id);
break;
case TYPE_CCE:
err = decode_cce(ac, &gb, ac->che[TYPE_SCE][elem_id]);
break;
case TYPE_LFE:
err = decode_ics(ac, &ac->che[TYPE_LFE][elem_id]->ch[0], &gb, 0, 0);
break;
case TYPE_DSE:
skip_data_stream_element(&gb);
err = 0;
break;
case TYPE_PCE:
{
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
if((err = decode_pce(ac, new_che_pos, &gb)))
break;
err = output_configure(ac, ac->che_pos, new_che_pos);
break;
}
case TYPE_FIL:
if (elem_id == 15)
elem_id += get_bits(&gb, 8) - 1;
while (elem_id > 0)
elem_id -= decode_extension_payload(ac, &gb, elem_id);
err = 0; /* FIXME */
break;
default:
err = -1; /* should not happen, but keeps compiler happy */
break;
}
if(err)
return err;
}
spectral_to_sample(ac);
if (!ac->is_saved) {
ac->is_saved = 1;
*data_size = 0;

46
libavcodec/aac.h
Unescape View File

@ -135,6 +135,17 @@ enum CouplingPoint {
/**
* Individual Channel Stream
*/
typedef struct {
uint8_t max_sfb; ///< number of scalefactor bands per group
enum WindowSequence window_sequence[2];
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];
const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
int num_swb; ///< number of scalefactor window bands
int num_windows;
int tns_max_bands;
} IndividualChannelStream;
/**
* Dynamic Range Control - decoded from the bitstream but not processed further.
@ -163,6 +174,41 @@ typedef struct {
* coupling parameters
*/
typedef struct {
enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied.
int num_coupled; ///< number of target elements
enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE.
int id_select[8]; ///< element id
int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for left channel;
* [2] list of gains for right channel; [3] lists of gains for both channels
*/
float gain[16][120];
} ChannelCoupling;
/**
* Single Channel Element - used for both SCE and LFE elements.
*/
typedef struct {
IndividualChannelStream ics;
TemporalNoiseShaping tns;
enum BandType band_type[120]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
DECLARE_ALIGNED_16(float, saved[1024]); ///< overlap
DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
} SingleChannelElement;
/**
* channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct {
// CPE specific
uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
// shared
SingleChannelElement ch[2];
// CCE specific
ChannelCoupling coup;
} ChannelElement;
/**
* main AAC context

Write Preview
Loading…
Cancel
Save