*/
/**
- * @file aac.c
+ * @file libavcodec/aac.c
* AAC decoder
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
* N (code in SoC repo) Long Term Prediction
* Y intensity stereo
* Y channel coupling
- * N frequency domain prediction
+ * Y frequency domain prediction
* Y Perceptual Noise Substitution
* Y Mid/Side stereo
* N Scalable Inverse AAC Quantization
#include "avcodec.h"
-#include "bitstream.h"
+#include "internal.h"
+#include "get_bits.h"
#include "dsputil.h"
+#include "lpc.h"
#include "aac.h"
#include "aactab.h"
#include "aacdectab.h"
#include "mpeg4audio.h"
+#include "aac_parser.h"
#include <assert.h>
#include <errno.h>
#include <math.h>
#include <string.h>
-#ifndef CONFIG_HARDCODED_TABLES
- static float ff_aac_ivquant_tab[IVQUANT_SIZE];
- static float ff_aac_pow2sf_tab[316];
-#endif /* CONFIG_HARDCODED_TABLES */
+union float754 {
+ float f;
+ uint32_t i;
+};
static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
+static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
+{
+ if (ac->tag_che_map[type][elem_id]) {
+ return ac->tag_che_map[type][elem_id];
+ }
+ if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {
+ return NULL;
+ }
+ switch (ac->m4ac.chan_config) {
+ case 7:
+ if (ac->tags_mapped == 3 && type == TYPE_CPE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
+ }
+ case 6:
+ /* 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] */
+ if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
+ }
+ case 5:
+ if (ac->tags_mapped == 2 && type == TYPE_CPE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
+ }
+ case 4:
+ if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
+ }
+ case 3:
+ case 2:
+ if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
+ } else if (ac->m4ac.chan_config == 2) {
+ return NULL;
+ }
+ case 1:
+ if (!ac->tags_mapped && type == TYPE_SCE) {
+ ac->tags_mapped++;
+ return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
+ }
+ default:
+ return NULL;
+ }
+}
+
+/**
+ * Check for the channel element in the current channel position configuration.
+ * If it exists, make sure the appropriate element is allocated and map the
+ * channel order to match the internal FFmpeg channel layout.
+ *
+ * @param che_pos current channel position configuration
+ * @param type channel element type
+ * @param id channel element id
+ * @param channels count of the number of channels in the configuration
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int che_configure(AACContext *ac,
+ enum ChannelPosition che_pos[4][MAX_ELEM_ID],
+ int type, int id,
+ int *channels)
+{
+ if (che_pos[type][id]) {
+ if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
+ return AVERROR(ENOMEM);
+ if (type != TYPE_CCE) {
+ ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
+ if (type == TYPE_CPE) {
+ ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
+ }
+ }
+ } else
+ av_freep(&ac->che[type][id]);
+ return 0;
+}
+
+/**
+ * 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],
+ int channel_config, enum OCStatus oc_type)
+{
+ AVCodecContext *avctx = ac->avccontext;
+ int i, type, channels = 0, ret;
+
+ memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+
+ if (channel_config) {
+ for (i = 0; i < tags_per_config[channel_config]; i++) {
+ if ((ret = che_configure(ac, che_pos,
+ aac_channel_layout_map[channel_config - 1][i][0],
+ aac_channel_layout_map[channel_config - 1][i][1],
+ &channels)))
+ return ret;
+ }
+
+ memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
+ ac->tags_mapped = 0;
+
+ avctx->channel_layout = aac_channel_layout[channel_config - 1];
+ } else {
+ /* 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:
+ * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
+ */
+
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ for (type = 0; type < 4; type++) {
+ if ((ret = che_configure(ac, che_pos, type, i, &channels)))
+ return ret;
+ }
+ }
+
+ memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
+ ac->tags_mapped = 4 * MAX_ELEM_ID;
+
+ avctx->channel_layout = 0;
+ }
+
+ avctx->channels = channels;
+
+ ac->output_configured = oc_type;
+
+ return 0;
+}
+
/**
* Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
*
* @param type speaker type/position for these channels
*/
static void decode_channel_map(enum ChannelPosition *cpe_map,
- enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) {
- while(n--) {
+ enum ChannelPosition *sce_map,
+ enum ChannelPosition type,
+ GetBitContext *gb, int n)
+{
+ while (n--) {
enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
map[get_bits(gb, 4)] = type;
}
*
* @return Returns error status. 0 - OK, !0 - error
*/
-static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
- GetBitContext * gb) {
- int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
+static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+ GetBitContext *gb)
+{
+ int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
skip_bits(gb, 2); // object_type
- ac->m4ac.sampling_index = get_bits(gb, 4);
- if(ac->m4ac.sampling_index > 11) {
- av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
- return -1;
- }
- ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];
+ sampling_index = get_bits(gb, 4);
+ if (ac->m4ac.sampling_index != sampling_index)
+ av_log(ac->avccontext, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
+
num_front = get_bits(gb, 4);
num_side = get_bits(gb, 4);
num_back = get_bits(gb, 4);
*
* @return Returns error status. 0 - OK, !0 - error
*/
-static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
- int channel_config)
+static int set_default_channel_config(AACContext *ac,
+ enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+ int channel_config)
{
- if(channel_config < 1 || channel_config > 7) {
+ if (channel_config < 1 || channel_config > 7) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
channel_config);
return -1;
* 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
*/
- if(channel_config != 2)
+ if (channel_config != 2)
new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
- if(channel_config > 1)
+ if (channel_config > 1)
new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
- if(channel_config == 4)
+ if (channel_config == 4)
new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center
- if(channel_config > 4)
+ if (channel_config > 4)
new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
- = AAC_CHANNEL_BACK; // back stereo
- if(channel_config > 5)
+ = AAC_CHANNEL_BACK; // back stereo
+ if (channel_config > 5)
new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE
- if(channel_config == 7)
+ if (channel_config == 7)
new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
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;
}
skip_bits(gb, 14); // coreCoderDelay
extension_flag = get_bits1(gb);
- if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||
- ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
+ if (ac->m4ac.object_type == AOT_AAC_SCALABLE ||
+ ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
skip_bits(gb, 3); // layerNr
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
if (channel_config == 0) {
skip_bits(gb, 4); // element_instance_tag
- if((ret = decode_pce(ac, new_che_pos, gb)))
+ if ((ret = decode_pce(ac, new_che_pos, gb)))
return ret;
} else {
- if((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
+ if ((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
return ret;
}
- if((ret = output_configure(ac, ac->che_pos, new_che_pos)))
+ if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
return ret;
if (extension_flag) {
switch (ac->m4ac.object_type) {
- case AOT_ER_BSAC:
- skip_bits(gb, 5); // numOfSubFrame
- skip_bits(gb, 11); // layer_length
- break;
- case AOT_ER_AAC_LC:
- case AOT_ER_AAC_LTP:
- case AOT_ER_AAC_SCALABLE:
- case AOT_ER_AAC_LD:
- skip_bits(gb, 3); /* aacSectionDataResilienceFlag
+ case AOT_ER_BSAC:
+ skip_bits(gb, 5); // numOfSubFrame
+ skip_bits(gb, 11); // layer_length
+ break;
+ case AOT_ER_AAC_LC:
+ case AOT_ER_AAC_LTP:
+ case AOT_ER_AAC_SCALABLE:
+ case AOT_ER_AAC_LD:
+ skip_bits(gb, 3); /* aacSectionDataResilienceFlag
* aacScalefactorDataResilienceFlag
* aacSpectralDataResilienceFlag
*/
- break;
+ break;
}
skip_bits1(gb); // extensionFlag3 (TBD in version 3)
}
*
* @return Returns error status. 0 - OK, !0 - error
*/
-static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) {
+static int decode_audio_specific_config(AACContext *ac, void *data,
+ int data_size)
+{
GetBitContext gb;
int i;
init_get_bits(&gb, data, data_size * 8);
- if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
+ if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
return -1;
- if(ac->m4ac.sampling_index > 11) {
+ if (ac->m4ac.sampling_index > 12) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
return -1;
}
skip_bits_long(&gb, i);
switch (ac->m4ac.object_type) {
+ case AOT_AAC_MAIN:
case AOT_AAC_LC:
if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
return -1;
return 0;
}
-static av_cold int aac_decode_init(AVCodecContext * avccontext) {
- AACContext * ac = avccontext->priv_data;
+/**
+ * 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 void reset_predict_state(PredictorState *ps)
+{
+ ps->r0 = 0.0f;
+ ps->r1 = 0.0f;
+ ps->cor0 = 0.0f;
+ ps->cor1 = 0.0f;
+ ps->var0 = 1.0f;
+ ps->var1 = 1.0f;
+}
+
+static void reset_all_predictors(PredictorState *ps)
+{
+ int i;
+ for (i = 0; i < MAX_PREDICTORS; i++)
+ reset_predict_state(&ps[i]);
+}
+
+static void reset_predictor_group(PredictorState *ps, int group_num)
+{
+ int i;
+ for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
+ reset_predict_state(&ps[i]);
+}
+
+static av_cold int aac_decode_init(AVCodecContext *avccontext)
+{
+ AACContext *ac = avccontext->priv_data;
int i;
ac->avccontext = avccontext;
- if (avccontext->extradata_size <= 0 ||
- decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
- return -1;
+ if (avccontext->extradata_size > 0) {
+ if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
+ return -1;
+ avccontext->sample_rate = ac->m4ac.sample_rate;
+ } else if (avccontext->channels > 0) {
+ ac->m4ac.sample_rate = avccontext->sample_rate;
+ }
- avccontext->sample_fmt = SAMPLE_FMT_S16;
- avccontext->sample_rate = ac->m4ac.sample_rate;
- avccontext->frame_size = 1024;
+ avccontext->sample_fmt = SAMPLE_FMT_S16;
+ avccontext->frame_size = 1024;
AAC_INIT_VLC_STATIC( 0, 144);
AAC_INIT_VLC_STATIC( 1, 114);
// 32768 - Required to scale values to the correct range for the bias method
// for float to int16 conversion.
- if(ac->dsp.float_to_int16 == ff_float_to_int16_c) {
- ac->add_bias = 385.0f;
- ac->sf_scale = 1. / (-1024. * 32768.);
+ if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
+ ac->add_bias = 385.0f;
+ ac->sf_scale = 1. / (-1024. * 32768.);
ac->sf_offset = 0;
} else {
- ac->add_bias = 0.0f;
- ac->sf_scale = 1. / -1024.;
+ ac->add_bias = 0.0f;
+ ac->sf_scale = 1. / -1024.;
ac->sf_offset = 60;
}
-#ifndef CONFIG_HARDCODED_TABLES
- for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++)
- ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] = cbrt(fabs(i)) * i;
- for (i = 0; i < 316; i++)
- ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
+#if !CONFIG_HARDCODED_TABLES
+ for (i = 0; i < 428; i++)
+ ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
#endif /* CONFIG_HARDCODED_TABLES */
- INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]),
- ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
- ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
- 352);
+ INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
+ ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
+ 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);
+ // 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);
+ ff_sine_window_init(ff_sine_1024, 1024);
+ ff_sine_window_init(ff_sine_128, 128);
- ff_mdct_init(&ac->mdct, 11, 1);
- ff_mdct_init(&ac->mdct_small, 8, 1);
return 0;
}
/**
* Skip data_stream_element; reference: table 4.10.
*/
-static void skip_data_stream_element(GetBitContext * gb) {
+static void skip_data_stream_element(GetBitContext *gb)
+{
int byte_align = get_bits1(gb);
int count = get_bits(gb, 8);
if (count == 255)
skip_bits_long(gb, 8 * count);
}
+static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
+ GetBitContext *gb)
+{
+ int sfb;
+ if (get_bits1(gb)) {
+ ics->predictor_reset_group = get_bits(gb, 5);
+ if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
+ return -1;
+ }
+ }
+ for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
+ ics->prediction_used[sfb] = get_bits1(gb);
+ }
+ return 0;
+}
+
/**
* Decode Individual Channel Stream info; reference: table 4.6.
*
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
*/
-static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) {
+static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
+ GetBitContext *gb, int common_window)
+{
if (get_bits1(gb)) {
av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
memset(ics, 0, sizeof(IndividualChannelStream));
}
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = get_bits(gb, 2);
- ics->use_kb_window[1] = ics->use_kb_window[0];
- ics->use_kb_window[0] = get_bits1(gb);
- ics->num_window_groups = 1;
- ics->group_len[0] = 1;
+ ics->use_kb_window[1] = ics->use_kb_window[0];
+ ics->use_kb_window[0] = get_bits1(gb);
+ ics->num_window_groups = 1;
+ ics->group_len[0] = 1;
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ int i;
+ ics->max_sfb = get_bits(gb, 4);
+ for (i = 0; i < 7; i++) {
+ if (get_bits1(gb)) {
+ ics->group_len[ics->num_window_groups - 1]++;
+ } else {
+ ics->num_window_groups++;
+ ics->group_len[ics->num_window_groups - 1] = 1;
+ }
+ }
+ ics->num_windows = 8;
+ ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index];
+ ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index];
+ ics->predictor_present = 0;
+ } else {
+ ics->max_sfb = get_bits(gb, 6);
+ ics->num_windows = 1;
+ ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index];
+ ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index];
+ ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
+ ics->predictor_present = get_bits1(gb);
+ ics->predictor_reset_group = 0;
+ if (ics->predictor_present) {
+ if (ac->m4ac.object_type == AOT_AAC_MAIN) {
+ if (decode_prediction(ac, ics, gb)) {
+ memset(ics, 0, sizeof(IndividualChannelStream));
+ return -1;
+ }
+ } else if (ac->m4ac.object_type == AOT_AAC_LC) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
+ memset(ics, 0, sizeof(IndividualChannelStream));
+ return -1;
+ } else {
+ av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
+ memset(ics, 0, sizeof(IndividualChannelStream));
+ return -1;
+ }
+ }
+ }
- return 0;
-}
+ 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;
+ }
-/**
- * inverse quantization
- *
- * @param a quantized value to be dequantized
- * @return Returns dequantized value.
- */
-static inline float ivquant(int a) {
- if (a + (unsigned int)IVQUANT_SIZE/2 - 1 < (unsigned int)IVQUANT_SIZE - 1)
- return ff_aac_ivquant_tab[a + IVQUANT_SIZE/2 - 1];
- else
- return cbrtf(fabsf(a)) * a;
+ return 0;
}
/**
*
* @return Returns error status. 0 - OK, !0 - error
*/
-static int decode_band_types(AACContext * ac, enum BandType band_type[120],
- int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
+static int decode_band_types(AACContext *ac, enum BandType band_type[120],
+ int band_type_run_end[120], GetBitContext *gb,
+ IndividualChannelStream *ics)
+{
int g, idx = 0;
const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
for (g = 0; g < ics->num_window_groups; g++) {
int k = 0;
while (k < ics->max_sfb) {
- uint8_t sect_len = k;
+ uint8_t sect_end = k;
int sect_len_incr;
int sect_band_type = get_bits(gb, 4);
if (sect_band_type == 12) {
av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
return -1;
}
- while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1)
- sect_len += sect_len_incr;
- sect_len += sect_len_incr;
- if (sect_len > ics->max_sfb) {
+ while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
+ sect_end += sect_len_incr;
+ sect_end += sect_len_incr;
+ if (sect_end > ics->max_sfb) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "Number of bands (%d) exceeds limit (%d).\n",
- sect_len, ics->max_sfb);
+ "Number of bands (%d) exceeds limit (%d).\n",
+ sect_end, ics->max_sfb);
return -1;
}
+ for (; k < sect_end; k++) {
+ band_type [idx] = sect_band_type;
+ band_type_run_end[idx++] = sect_end;
+ }
}
}
return 0;
/**
* Decode scalefactors; reference: table 4.47.
*
- * @param mix_gain channel gain (Not used by AAC bitstream.)
* @param global_gain first scalefactor value as scalefactors are differentially coded
* @param band_type array of the used band type
* @param band_type_run_end array of the last scalefactor band of a band type run
*
* @return Returns error status. 0 - OK, !0 - error
*/
-static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb,
- float mix_gain, unsigned int global_gain, IndividualChannelStream * ics,
- enum BandType band_type[120], int band_type_run_end[120]) {
+static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
+ unsigned int global_gain,
+ IndividualChannelStream *ics,
+ enum BandType band_type[120],
+ int band_type_run_end[120])
+{
const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
int g, i, idx = 0;
int offset[3] = { global_gain, global_gain - 90, 100 };
int noise_flag = 1;
static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
- ics->intensity_present = 0;
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb;) {
int run_end = band_type_run_end[idx];
if (band_type[idx] == ZERO_BT) {
- for(; i < run_end; i++, idx++)
+ for (; i < run_end; i++, idx++)
sf[idx] = 0.;
- }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
- ics->intensity_present = 1;
- for(; i < run_end; i++, idx++) {
+ } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
+ for (; i < run_end; i++, idx++) {
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
- if(offset[2] > 255U) {
+ if (offset[2] > 255U) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "%s (%d) out of range.\n", sf_str[2], offset[2]);
+ "%s (%d) out of range.\n", sf_str[2], offset[2]);
return -1;
}
- sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
- sf[idx] *= mix_gain;
+ sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
}
- }else if(band_type[idx] == NOISE_BT) {
- for(; i < run_end; i++, idx++) {
- if(noise_flag-- > 0)
+ } else if (band_type[idx] == NOISE_BT) {
+ for (; i < run_end; i++, idx++) {
+ if (noise_flag-- > 0)
offset[1] += get_bits(gb, 9) - 256;
else
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
- if(offset[1] > 255U) {
+ if (offset[1] > 255U) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "%s (%d) out of range.\n", sf_str[1], offset[1]);
+ "%s (%d) out of range.\n", sf_str[1], offset[1]);
return -1;
}
- sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset];
- sf[idx] *= mix_gain;
+ sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
}
- }else {
- for(; i < run_end; i++, idx++) {
+ } else {
+ for (; i < run_end; i++, idx++) {
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
- if(offset[0] > 255U) {
+ if (offset[0] > 255U) {
av_log(ac->avccontext, AV_LOG_ERROR,
- "%s (%d) out of range.\n", sf_str[0], offset[0]);
+ "%s (%d) out of range.\n", sf_str[0], offset[0]);
return -1;
}
sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
- sf[idx] *= mix_gain;
}
}
}
/**
* Decode pulse data; reference: table 4.7.
*/
-static void decode_pulses(Pulse * pulse, GetBitContext * gb) {
- int i;
+static int decode_pulses(Pulse *pulse, GetBitContext *gb,
+ const uint16_t *swb_offset, int num_swb)
+{
+ int i, pulse_swb;
pulse->num_pulse = get_bits(gb, 2) + 1;
- pulse->start = get_bits(gb, 6);
- for (i = 0; i < pulse->num_pulse; i++) {
- pulse->offset[i] = get_bits(gb, 5);
- pulse->amp [i] = get_bits(gb, 4);
+ pulse_swb = get_bits(gb, 6);
+ if (pulse_swb >= num_swb)
+ return -1;
+ pulse->pos[0] = swb_offset[pulse_swb];
+ pulse->pos[0] += get_bits(gb, 5);
+ if (pulse->pos[0] > 1023)
+ return -1;
+ pulse->amp[0] = get_bits(gb, 4);
+ for (i = 1; i < pulse->num_pulse; i++) {
+ pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
+ if (pulse->pos[i] > 1023)
+ return -1;
+ pulse->amp[i] = get_bits(gb, 4);
+ }
+ return 0;
+}
+
+/**
+ * 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, coef_compress;
+ 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++) {
+ if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
+ coef_res = get_bits1(gb);
+
+ for (filt = 0; filt < tns->n_filt[w]; filt++) {
+ int tmp2_idx;
+ tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
+
+ if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
+ 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->order[w][filt]) {
+ tns->direction[w][filt] = get_bits1(gb);
+ coef_compress = get_bits1(gb);
+ coef_len = coef_res + 3 - coef_compress;
+ tmp2_idx = 2 * coef_compress + coef_res;
+
+ for (i = 0; i < tns->order[w][filt]; i++)
+ tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
+ }
+ }
+ }
}
+ return 0;
}
/**
* [1] mask is decoded from bitstream; [2] mask is all 1s;
* [3] reserved for scalable AAC
*/
-static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
- int ms_present) {
+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.
+ * Decode spectral data; reference: table 4.50.
+ * Dequantize and scale spectral data; reference: 4.6.3.3.
+ *
+ * @param coef array of dequantized, scaled spectral data
+ * @param sf array of scalefactors or intensity stereo positions
+ * @param pulse_present set if pulses are present
+ * @param pulse pointer to pulse data struct
+ * @param band_type array of the used band type
*
- * @param pulse pointer to pulse data struct
- * @param icoef array of quantized spectral data
+ * @return Returns error status. 0 - OK, !0 - error
*/
-static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) {
- int i, off = ics->swb_offset[pulse->start];
- for (i = 0; i < pulse->num_pulse; i++) {
- int ic;
- off += pulse->offset[i];
- ic = (icoef[off] - 1)>>31;
- icoef[off] += (pulse->amp[i]^ic) - ic;
+static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
+ GetBitContext *gb, float sf[120],
+ int pulse_present, const Pulse *pulse,
+ const IndividualChannelStream *ics,
+ enum BandType band_type[120])
+{
+ int i, k, g, idx = 0;
+ const int c = 1024 / ics->num_windows;
+ const uint16_t *offsets = ics->swb_offset;
+ float *coef_base = coef;
+ static const float sign_lookup[] = { 1.0f, -1.0f };
+
+ for (g = 0; g < ics->num_windows; g++)
+ memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
+
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ const int cur_band_type = band_type[idx];
+ const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4;
+ const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type);
+ int group;
+ if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) {
+ for (group = 0; group < ics->group_len[g]; group++) {
+ memset(coef + group * 128 + offsets[i], 0, (offsets[i + 1] - offsets[i]) * sizeof(float));
+ }
+ } else if (cur_band_type == NOISE_BT) {
+ for (group = 0; group < ics->group_len[g]; group++) {
+ float scale;
+ float band_energy;
+ float *cf = coef + group * 128 + offsets[i];
+ int len = offsets[i+1] - offsets[i];
+
+ for (k = 0; k < len; k++) {
+ ac->random_state = lcg_random(ac->random_state);
+ cf[k] = ac->random_state;
+ }
+
+ band_energy = ac->dsp.scalarproduct_float(cf, cf, len);
+ scale = sf[idx] / sqrtf(band_energy);
+ ac->dsp.vector_fmul_scalar(cf, cf, scale, len);
+ }
+ } else {
+ for (group = 0; group < ics->group_len[g]; group++) {
+ const float *vq[96];
+ const float **vqp = vq;
+ float *cf = coef + (group << 7) + offsets[i];
+ int len = offsets[i + 1] - offsets[i];
+
+ for (k = offsets[i]; k < offsets[i + 1]; k += dim) {
+ const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3);
+ const int coef_tmp_idx = (group << 7) + k;
+ const float *vq_ptr;
+ int j;
+ if (index >= ff_aac_spectral_sizes[cur_band_type - 1]) {
+ av_log(ac->avccontext, AV_LOG_ERROR,
+ "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
+ cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]);
+ return -1;
+ }
+ vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim];
+ *vqp++ = vq_ptr;
+ if (is_cb_unsigned) {
+ if (vq_ptr[0])
+ coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)];
+ if (vq_ptr[1])
+ coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)];
+ if (dim == 4) {
+ if (vq_ptr[2])
+ coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)];
+ if (vq_ptr[3])
+ coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)];
+ }
+ if (cur_band_type == ESC_BT) {
+ for (j = 0; j < 2; j++) {
+ if (vq_ptr[j] == 64.0f) {
+ int n = 4;
+ /* The total length of escape_sequence must be < 22 bits according
+ to the specification (i.e. max is 11111111110xxxxxxxxxx). */
+ while (get_bits1(gb) && n < 15) n++;
+ if (n == 15) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
+ return -1;
+ }
+ n = (1 << n) + get_bits(gb, n);
+ coef[coef_tmp_idx + j] *= cbrtf(n) * n;
+ } else
+ coef[coef_tmp_idx + j] *= vq_ptr[j];
+ }
+ }
+ }
+ }
+
+ if (is_cb_unsigned && cur_band_type != ESC_BT) {
+ ac->dsp.vector_fmul_sv_scalar[dim>>2](
+ cf, cf, vq, sf[idx], len);
+ } else if (cur_band_type == ESC_BT) {
+ ac->dsp.vector_fmul_scalar(cf, cf, sf[idx], len);
+ } else { /* !is_cb_unsigned */
+ ac->dsp.sv_fmul_scalar[dim>>2](cf, vq, sf[idx], len);
+ }
+ }
+ }
+ }
+ coef += ics->group_len[g] << 7;
+ }
+
+ if (pulse_present) {
+ idx = 0;
+ for (i = 0; i < pulse->num_pulse; i++) {
+ float co = coef_base[ pulse->pos[i] ];
+ while (offsets[idx + 1] <= pulse->pos[i])
+ idx++;
+ if (band_type[idx] != NOISE_BT && sf[idx]) {
+ float ico = -pulse->amp[i];
+ if (co) {
+ co /= sf[idx];
+ ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
+ }
+ coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
+ }
+ }
}
+ return 0;
+}
+
+static av_always_inline float flt16_round(float pf)
+{
+ union float754 tmp;
+ tmp.f = pf;
+ tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
+ return tmp.f;
+}
+
+static av_always_inline float flt16_even(float pf)
+{
+ union float754 tmp;
+ tmp.f = pf;
+ tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
+ return tmp.f;
+}
+
+static av_always_inline float flt16_trunc(float pf)
+{
+ union float754 pun;
+ pun.f = pf;
+ pun.i &= 0xFFFF0000U;
+ return pun.f;
+}
+
+static void predict(AACContext *ac, PredictorState *ps, float *coef,
+ int output_enable)
+{
+ const float a = 0.953125; // 61.0 / 64
+ const float alpha = 0.90625; // 29.0 / 32
+ float e0, e1;
+ float pv;
+ float k1, k2;
+
+ k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
+ k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
+
+ pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
+ if (output_enable)
+ *coef += pv * ac->sf_scale;
+
+ e0 = *coef / ac->sf_scale;
+ e1 = e0 - k1 * ps->r0;
+
+ ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
+ ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
+ ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
+ ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
+
+ ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
+ ps->r0 = flt16_trunc(a * e0);
+}
+
+/**
+ * Apply AAC-Main style frequency domain prediction.
+ */
+static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
+{
+ int sfb, k;
+
+ if (!sce->ics.predictor_initialized) {
+ reset_all_predictors(sce->predictor_state);
+ sce->ics.predictor_initialized = 1;
+ }
+
+ if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
+ for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
+ for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
+ predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
+ sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
+ }
+ }
+ if (sce->ics.predictor_reset_group)
+ reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
+ } else
+ reset_all_predictors(sce->predictor_state);
}
/**
*
* @return Returns error status. 0 - OK, !0 - error
*/
-static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
- int icoeffs[1024];
+static int decode_ics(AACContext *ac, SingleChannelElement *sce,
+ GetBitContext *gb, int common_window, int scale_flag)
+{
Pulse pulse;
- TemporalNoiseShaping * tns = &sce->tns;
- IndividualChannelStream * ics = &sce->ics;
- float * out = sce->coeffs;
+ TemporalNoiseShaping *tns = &sce->tns;
+ IndividualChannelStream *ics = &sce->ics;
+ float *out = sce->coeffs;
int global_gain, pulse_present = 0;
- /* These two assignments are to silence some GCC warnings about the
- * variables being used uninitialised when in fact they always are.
+ /* This assignment is to silence a GCC warning about the variable being used
+ * uninitialized when in fact it always is.
*/
pulse.num_pulse = 0;
- pulse.start = 0;
global_gain = get_bits(gb, 8);
av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
return -1;
}
- decode_pulses(&pulse, gb);
+ if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
+ return -1;
+ }
}
if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
return -1;
}
}
- if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0)
+ if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
return -1;
- if (pulse_present)
- add_pulses(icoeffs, &pulse, ics);
- dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type);
+
+ if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
+ apply_prediction(ac, sce);
+
return 0;
}
+/**
+ * Mid/Side stereo decoding; reference: 4.6.8.1.3.
+ */
+static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
+{
+ const IndividualChannelStream *ics = &cpe->ch[0].ics;
+ float *ch0 = cpe->ch[0].coeffs;
+ float *ch1 = cpe->ch[1].coeffs;
+ int g, i, group, idx = 0;
+ const uint16_t *offsets = ics->swb_offset;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb; i++, idx++) {
+ if (cpe->ms_mask[idx] &&
+ cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
+ for (group = 0; group < ics->group_len[g]; group++) {
+ ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
+ ch1 + group * 128 + offsets[i],
+ offsets[i+1] - offsets[i]);
+ }
+ }
+ }
+ ch0 += ics->group_len[g] * 128;
+ ch1 += ics->group_len[g] * 128;
+ }
+}
+
+/**
+ * intensity stereo decoding; reference: 4.6.8.2.3
+ *
+ * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
+ * [1] mask is decoded from bitstream; [2] mask is all 1s;
+ * [3] reserved for scalable AAC
+ */
+static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
+{
+ const IndividualChannelStream *ics = &cpe->ch[1].ics;
+ SingleChannelElement *sce1 = &cpe->ch[1];
+ float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
+ const uint16_t *offsets = ics->swb_offset;
+ int g, group, i, k, idx = 0;
+ int c;
+ float scale;
+ for (g = 0; g < ics->num_window_groups; g++) {
+ for (i = 0; i < ics->max_sfb;) {
+ if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
+ const int bt_run_end = sce1->band_type_run_end[idx];
+ for (; i < bt_run_end; i++, idx++) {
+ c = -1 + 2 * (sce1->band_type[idx] - 14);
+ if (ms_present)
+ c *= 1 - 2 * cpe->ms_mask[idx];
+ scale = c * sce1->sf[idx];
+ for (group = 0; group < ics->group_len[g]; group++)
+ for (k = offsets[i]; k < offsets[i + 1]; k++)
+ coef1[group * 128 + k] = scale * coef0[group * 128 + k];
+ }
+ } else {
+ int bt_run_end = sce1->band_type_run_end[idx];
+ idx += bt_run_end - i;
+ i = bt_run_end;
+ }
+ }
+ coef0 += ics->group_len[g] * 128;
+ coef1 += ics->group_len[g] * 128;
+ }
+}
+
/**
* Decode a channel_pair_element; reference: table 4.4.
*
*
* @return Returns error status. 0 - OK, !0 - error
*/
-static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
+static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
+{
int i, ret, common_window, ms_present = 0;
- ChannelElement * cpe;
- cpe = ac->che[TYPE_CPE][elem_id];
common_window = get_bits1(gb);
if (common_window) {
if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
cpe->ch[1].ics = cpe->ch[0].ics;
cpe->ch[1].ics.use_kb_window[1] = i;
ms_present = get_bits(gb, 2);
- if(ms_present == 3) {
+ if (ms_present == 3) {
av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
return -1;
- } else if(ms_present)
+ } else if (ms_present)
decode_mid_side_stereo(cpe, gb, ms_present);
}
if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
return ret;
- if (common_window && ms_present)
- apply_mid_side_stereo(cpe);
+ if (common_window) {
+ if (ms_present)
+ apply_mid_side_stereo(ac, cpe);
+ if (ac->m4ac.object_type == AOT_AAC_MAIN) {
+ apply_prediction(ac, &cpe->ch[0]);
+ apply_prediction(ac, &cpe->ch[1]);
+ }
+ }
+
+ apply_intensity_stereo(cpe, ms_present);
+ return 0;
+}
- if (cpe->ch[1].ics.intensity_present)
- apply_intensity_stereo(cpe, ms_present);
+/**
+ * Decode coupling_channel_element; reference: table 4.8.
+ *
+ * @param elem_id Identifies the instance of a syntax element.
+ *
+ * @return Returns error status. 0 - OK, !0 - error
+ */
+static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
+{
+ int num_gain = 0;
+ int c, g, sfb, ret;
+ int sign;
+ float scale;
+ SingleChannelElement *sce = &che->ch[0];
+ ChannelCoupling *coup = &che->coup;
+
+ 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] = 2;
+ }
+ coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
+
+ sign = get_bits(gb, 1);
+ scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
+
+ if ((ret = decode_ics(ac, sce, gb, 0, 0)))
+ return ret;
+
+ for (c = 0; c < num_gain; c++) {
+ int idx = 0;
+ 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);
+ }
+ if (coup->coupling_point == AFTER_IMDCT) {
+ coup->gain[c][0] = gain_cache;
+ } else {
+ 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;
+ t = gain += t;
+ if (sign) {
+ s -= 2 * (t & 0x1);
+ t >>= 1;
+ }
+ gain_cache = pow(scale, -t) * s;
+ }
+ }
+ coup->gain[c][idx] = gain_cache;
+ }
+ }
+ }
+ }
+ }
return 0;
}
*
* @return Returns number of bytes consumed from the TYPE_FIL element.
*/
-static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
+static int decode_sbr_extension(AACContext *ac, GetBitContext *gb,
+ int crc, int cnt)
+{
// TODO : sbr_extension implementation
- av_log(ac->avccontext, AV_LOG_DEBUG, "aac: SBR not yet supported.\n");
- skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type
+ av_log_missing_feature(ac->avccontext, "SBR", 0);
+ skip_bits_long(gb, 8 * cnt - 4); // -4 due to reading extension type
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.
*
*
* @return Returns number of bytes consumed.
*/
-static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {
- int n = 1;
+static int decode_dynamic_range(DynamicRangeControl *che_drc,
+ GetBitContext *gb, int cnt)
+{
+ int n = 1;
int drc_num_bands = 1;
int i;
/* pce_tag_present? */
- if(get_bits1(gb)) {
+ if (get_bits1(gb)) {
che_drc->pce_instance_tag = get_bits(gb, 4);
skip_bits(gb, 4); // tag_reserved_bits
n++;
}
/* excluded_chns_present? */
- if(get_bits1(gb)) {
+ if (get_bits1(gb)) {
n += decode_drc_channel_exclusions(che_drc, gb);
}
*
* @return Returns number of bytes consumed
*/
-static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {
+static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt)
+{
int crc_flag = 0;
int res = cnt;
switch (get_bits(gb, 4)) { // extension type
- case EXT_SBR_DATA_CRC:
- crc_flag++;
- case EXT_SBR_DATA:
- res = decode_sbr_extension(ac, gb, crc_flag, cnt);
- break;
- case EXT_DYNAMIC_RANGE:
- res = decode_dynamic_range(&ac->che_drc, gb, cnt);
- break;
- case EXT_FILL:
- case EXT_FILL_DATA:
- case EXT_DATA_ELEMENT:
- default:
- skip_bits_long(gb, 8*cnt - 4);
- break;
+ case EXT_SBR_DATA_CRC:
+ crc_flag++;
+ case EXT_SBR_DATA:
+ res = decode_sbr_extension(ac, gb, crc_flag, cnt);
+ break;
+ case EXT_DYNAMIC_RANGE:
+ res = decode_dynamic_range(&ac->che_drc, gb, cnt);
+ break;
+ case EXT_FILL:
+ case EXT_FILL_DATA:
+ case EXT_DATA_ELEMENT:
+ default:
+ skip_bits_long(gb, 8 * cnt - 4);
+ break;
};
return res;
}
+/**
+ * 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;
+ int bottom, top, order, start, end, size, inc;
+ float lpc[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 = tns->order[w][filt];
+ if (order == 0)
+ continue;
+
+ // tns_decode_coef
+ compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
+
+ 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
+ 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];
+ }
+ }
+}
+
/**
* 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;
- const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
- const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
- const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
- const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
- float * buf = ac->buf_mdct;
+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;
+ 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;
+ float *temp = ac->temp;
int i;
+ // imdct
+ 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 < 1024; i += 128)
+ ff_imdct_half(&ac->mdct_small, buf + i, in + i);
+ } else
+ ff_imdct_half(&ac->mdct, buf, in);
+
+ /* 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
+ * transitions. This leaves just two cases (long to long and short to short)
+ * with a little special sauce for EIGHT_SHORT_SEQUENCE.
+ */
+ if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
+ (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
+ ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512);
+ } else {
+ for (i = 0; i < 448; i++)
+ out[i] = saved[i] + ac->add_bias;
+
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, ac->add_bias, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, ac->add_bias, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, ac->add_bias, 64);
+ ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64);
+ memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
+ } else {
+ ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64);
+ for (i = 576; i < 1024; i++)
+ out[i] = buf[i-512] + ac->add_bias;
+ }
+ }
+
+ // buffer update
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ for (i = 0; i < 64; i++)
+ saved[i] = temp[64 + i] - ac->add_bias;
+ ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
+ 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));
+ } 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));
+ } else { // LONG_STOP or ONLY_LONG
+ memcpy( saved, buf + 512, 512 * sizeof(float));
+ }
+}
+
/**
* Apply dependent channel coupling (applied before IMDCT).
*
* @param index index into coupling gain array
*/
-static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
- IndividualChannelStream * ics = &cc->ch[0].ics;
- const uint16_t * offsets = ics->swb_offset;
- float * dest = sce->coeffs;
- const float * src = cc->ch[0].coeffs;
+static void apply_dependent_coupling(AACContext *ac,
+ SingleChannelElement *target,
+ ChannelElement *cce, int index)
+{
+ IndividualChannelStream *ics = &cce->ch[0].ics;
+ const uint16_t *offsets = ics->swb_offset;
+ float *dest = target->coeffs;
+ const float *src = cce->ch[0].coeffs;
int g, i, group, k, idx = 0;
- if(ac->m4ac.object_type == AOT_AAC_LTP) {
+ if (ac->m4ac.object_type == AOT_AAC_LTP) {
av_log(ac->avccontext, AV_LOG_ERROR,
"Dependent coupling is not supported together with LTP\n");
return;
}
for (g = 0; g < ics->num_window_groups; g++) {
for (i = 0; i < ics->max_sfb; i++, idx++) {
- if (cc->ch[0].band_type[idx] != ZERO_BT) {
- float gain = cc->coup.gain[index][idx] * sce->mixing_gain;
+ if (cce->ch[0].band_type[idx] != ZERO_BT) {
+ const float gain = cce->coup.gain[index][idx];
for (group = 0; group < ics->group_len[g]; group++) {
- for (k = offsets[i]; k < offsets[i+1]; k++) {
+ for (k = offsets[i]; k < offsets[i + 1]; k++) {
// XXX dsputil-ize
- dest[group*128+k] += gain * src[group*128+k];
+ dest[group * 128 + k] += gain * src[group * 128 + k];
}
}
}
}
- dest += ics->group_len[g]*128;
- src += ics->group_len[g]*128;
+ dest += ics->group_len[g] * 128;
+ src += ics->group_len[g] * 128;
}
}
*
* @param index index into coupling gain array
*/
-static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
+static void apply_independent_coupling(AACContext *ac,
+ SingleChannelElement *target,
+ ChannelElement *cce, int index)
+{
int i;
- float gain = cc->coup.gain[index][0] * sce->mixing_gain;
+ const float gain = cce->coup.gain[index][0];
+ const float bias = ac->add_bias;
+ const float *src = cce->ch[0].ret;
+ float *dest = target->ret;
+
for (i = 0; i < 1024; i++)
- sce->ret[i] += gain * (cc->ch[0].ret[i] - ac->add_bias);
+ dest[i] += gain * (src[i] - bias);
+}
+
+/**
+ * channel coupling transformation interface
+ *
+ * @param index index into coupling gain array
+ * @param apply_coupling_method pointer to (in)dependent coupling function
+ */
+static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
+ enum RawDataBlockType type, int elem_id,
+ enum CouplingPoint coupling_point,
+ void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
+{
+ int i, c;
+
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ ChannelElement *cce = ac->che[TYPE_CCE][i];
+ int index = 0;
+
+ if (cce && cce->coup.coupling_point == coupling_point) {
+ ChannelCoupling *coup = &cce->coup;
+
+ for (c = 0; c <= coup->num_coupled; c++) {
+ if (coup->type[c] == type && coup->id_select[c] == elem_id) {
+ if (coup->ch_select[c] != 1) {
+ apply_coupling_method(ac, &cc->ch[0], cce, index);
+ if (coup->ch_select[c] != 0)
+ index++;
+ }
+ if (coup->ch_select[c] != 2)
+ apply_coupling_method(ac, &cc->ch[1], cce, index++);
+ } else
+ index += 1 + (coup->ch_select[c] == 3);
+ }
+ }
+ }
}
+/**
+ * Convert spectral data to float samples, applying all supported tools as appropriate.
+ */
+static void spectral_to_sample(AACContext *ac)
+{
+ int i, type;
+ for (type = 3; type >= 0; type--) {
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ ChannelElement *che = ac->che[type][i];
+ if (che) {
+ if (type <= TYPE_CPE)
+ apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
+ 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)
+ apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
+ if (type <= TYPE_CPE)
+ apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
+ if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT)
+ imdct_and_windowing(ac, &che->ch[0]);
+ if (type == TYPE_CPE)
+ imdct_and_windowing(ac, &che->ch[1]);
+ if (type <= TYPE_CCE)
+ apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
+ }
+ }
+ }
+}
+
+static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
+{
+ int size;
+ AACADTSHeaderInfo hdr_info;
+
+ size = ff_aac_parse_header(gb, &hdr_info);
+ if (size > 0) {
+ if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
+ enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
+ memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
+ ac->m4ac.chan_config = hdr_info.chan_config;
+ if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config))
+ return -7;
+ if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
+ return -7;
+ } else if (ac->output_configured != OC_LOCKED) {
+ ac->output_configured = OC_NONE;
+ }
+ if (ac->output_configured != OC_LOCKED)
+ ac->m4ac.sbr = -1;
+ ac->m4ac.sample_rate = hdr_info.sample_rate;
+ ac->m4ac.sampling_index = hdr_info.sampling_index;
+ ac->m4ac.object_type = hdr_info.object_type;
+ if (!ac->avccontext->sample_rate)
+ ac->avccontext->sample_rate = hdr_info.sample_rate;
+ if (hdr_info.num_aac_frames == 1) {
+ if (!hdr_info.crc_absent)
+ skip_bits(gb, 16);
+ } else {
+ av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
+ return -1;
+ }
+ }
+ return size;
+}
+
+static int aac_decode_frame(AVCodecContext *avccontext, void *data,
+ int *data_size, AVPacket *avpkt)
+{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ AACContext *ac = avccontext->priv_data;
+ ChannelElement *che = NULL;
+ GetBitContext gb;
+ enum RawDataBlockType elem_type;
+ int err, elem_id, data_size_tmp;
+
+ init_get_bits(&gb, buf, buf_size * 8);
+
+ if (show_bits(&gb, 12) == 0xfff) {
+ if (parse_adts_frame_header(ac, &gb) < 0) {
+ av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
+ return -1;
+ }
+ if (ac->m4ac.sampling_index > 12) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
+ return -1;
+ }
+ }
+
+ // parse
+ while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
+ elem_id = get_bits(&gb, 4);
+
+ if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
+ av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
+ return -1;
+ }
+
+ switch (elem_type) {
+
+ case TYPE_SCE:
+ err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
+ break;
+
+ case TYPE_CPE:
+ err = decode_cpe(ac, &gb, che);
+ break;
+
+ case TYPE_CCE:
+ err = decode_cce(ac, &gb, che);
+ break;
+
+ case TYPE_LFE:
+ err = decode_ics(ac, &che->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;
+ if (ac->output_configured > OC_TRIAL_PCE)
+ av_log(avccontext, AV_LOG_ERROR,
+ "Not evaluating a further program_config_element as this construct is dubious at best.\n");
+ else
+ err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
+ 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;
- return 0;
+ return buf_size;
}
data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
- if(*data_size < data_size_tmp) {
+ if (*data_size < data_size_tmp) {
av_log(avccontext, AV_LOG_ERROR,
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
*data_size, data_size_tmp);
ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
+ if (ac->output_configured)
+ ac->output_configured = OC_LOCKED;
+
return buf_size;
}
-static av_cold int aac_decode_close(AVCodecContext * avccontext) {
- AACContext * ac = avccontext->priv_data;
- int i, j;
+static av_cold int aac_decode_close(AVCodecContext *avccontext)
+{
+ AACContext *ac = avccontext->priv_data;
+ int i, type;
for (i = 0; i < MAX_ELEM_ID; i++) {
- for(j = 0; j < 4; j++)
- av_freep(&ac->che[j][i]);
+ for (type = 0; type < 4; type++)
+ av_freep(&ac->che[type][i]);
}
ff_mdct_end(&ac->mdct);
ff_mdct_end(&ac->mdct_small);
- return 0 ;
+ return 0;
}
AVCodec aac_decoder = {
aac_decode_close,
aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
- .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+ .sample_fmts = (const enum SampleFormat[]) {
+ SAMPLE_FMT_S16,SAMPLE_FMT_NONE
+ },
+ .channel_layouts = aac_channel_layout,
};
projects / ffmpeg / blobdiff