* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-ffmpeg@jannau.net>
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
* Y filterbank - standard
* N (code in SoC repo) filterbank - Scalable Sample Rate
* Y Temporal Noise Shaping
- * N (code in SoC repo) Long Term Prediction
+ * Y Long Term Prediction
* Y intensity stereo
* Y channel coupling
* Y frequency domain prediction
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
+#include "fmtconvert.h"
#include "lpc.h"
+#include "kbdwin.h"
+#include "sinewin.h"
#include "aac.h"
#include "aactab.h"
#include "aacsbr.h"
#include "mpeg4audio.h"
#include "aacadtsdec.h"
+#include "libavutil/intfloat.h"
#include <assert.h>
#include <errno.h>
# include "arm/aac.h"
#endif
-union float754 {
- float f;
- uint32_t i;
-};
-
static VLC vlc_scalefactors;
static VLC vlc_spectral[11];
}
}
+static int count_channels(enum ChannelPosition che_pos[4][MAX_ELEM_ID])
+{
+ int i, type, sum = 0;
+ for (i = 0; i < MAX_ELEM_ID; i++) {
+ for (type = 0; type < 4; type++) {
+ sum += (1 + (type == TYPE_CPE)) *
+ (che_pos[type][i] != AAC_CHANNEL_OFF &&
+ che_pos[type][i] != AAC_CHANNEL_CC);
+ }
+ }
+ return sum;
+}
+
/**
* 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.
+ * channel order to match the internal Libav channel layout.
*
* @param che_pos current channel position configuration
* @param type channel element type
* @return Returns error status. 0 - OK, !0 - error
*/
static av_cold int che_configure(AACContext *ac,
- enum ChannelPosition che_pos[4][MAX_ELEM_ID],
- int type, int id,
- int *channels)
+ 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);
- ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
+ if (!ac->che[type][id]) {
+ if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
+ return AVERROR(ENOMEM);
+ ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
+ }
if (type != TYPE_CCE) {
ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
if (type == TYPE_CPE ||
* @return Returns error status. 0 - OK, !0 - error
*/
static av_cold 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)
+ 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->avctx;
int i, type, channels = 0, ret;
return ret;
}
- memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
+ memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
avctx->channel_layout = aac_channel_layout[channel_config - 1];
} else {
* @return Returns error status. 0 - OK, !0 - error
*/
static av_cold int set_default_channel_config(AVCodecContext *avctx,
- enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
- int channel_config)
+ enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
+ int channel_config)
{
if (channel_config < 1 || channel_config > 7) {
av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
return ret;
}
+
+ if (count_channels(new_che_pos) > 1) {
+ m4ac->ps = 0;
+ } else if (m4ac->sbr == 1 && m4ac->ps == -1)
+ m4ac->ps = 1;
+
if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
return ret;
* @param ac pointer to AACContext, may be null
* @param avctx pointer to AVCCodecContext, used for logging
* @param m4ac pointer to MPEG4AudioConfig, used for parsing
- * @param data pointer to AVCodecContext extradata
- * @param data_size size of AVCCodecContext extradata
+ * @param data pointer to buffer holding an audio specific config
+ * @param bit_size size of audio specific config or data in bits
+ * @param sync_extension look for an appended sync extension
*
* @return Returns error status or number of consumed bits. <0 - error
*/
static int decode_audio_specific_config(AACContext *ac,
AVCodecContext *avctx,
MPEG4AudioConfig *m4ac,
- const uint8_t *data, int data_size)
+ const uint8_t *data, int bit_size,
+ int sync_extension)
{
GetBitContext gb;
int i;
- init_get_bits(&gb, data, data_size * 8);
+ av_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
+ for (i = 0; i < avctx->extradata_size; i++)
+ av_dlog(avctx, "%02x ", avctx->extradata[i]);
+ av_dlog(avctx, "\n");
- if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
+ init_get_bits(&gb, data, bit_size);
+
+ if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size, sync_extension)) < 0)
return -1;
if (m4ac->sampling_index > 12) {
av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
return -1;
}
- if (m4ac->sbr == 1 && m4ac->ps == -1)
- m4ac->ps = 1;
skip_bits_long(&gb, i);
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;
return -1;
}
+ av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
+ m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
+ m4ac->sample_rate, m4ac->sbr, m4ac->ps);
+
return get_bits_count(&gb);
}
reset_predict_state(&ps[i]);
}
+static int sample_rate_idx (int rate)
+{
+ if (92017 <= rate) return 0;
+ else if (75132 <= rate) return 1;
+ else if (55426 <= rate) return 2;
+ else if (46009 <= rate) return 3;
+ else if (37566 <= rate) return 4;
+ else if (27713 <= rate) return 5;
+ else if (23004 <= rate) return 6;
+ else if (18783 <= rate) return 7;
+ else if (13856 <= rate) return 8;
+ else if (11502 <= rate) return 9;
+ else if (9391 <= rate) return 10;
+ else return 11;
+}
+
static void reset_predictor_group(PredictorState *ps, int group_num)
{
int i;
static av_cold int aac_decode_init(AVCodecContext *avctx)
{
AACContext *ac = avctx->priv_data;
+ float output_scale_factor;
ac->avctx = avctx;
ac->m4ac.sample_rate = avctx->sample_rate;
if (avctx->extradata_size > 0) {
if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
avctx->extradata,
- avctx->extradata_size) < 0)
+ avctx->extradata_size*8, 1) < 0)
return -1;
+ } else {
+ int sr, i;
+ enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
+
+ sr = sample_rate_idx(avctx->sample_rate);
+ ac->m4ac.sampling_index = sr;
+ ac->m4ac.channels = avctx->channels;
+ ac->m4ac.sbr = -1;
+ ac->m4ac.ps = -1;
+
+ for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
+ if (ff_mpeg4audio_channels[i] == avctx->channels)
+ break;
+ if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
+ i = 0;
+ }
+ ac->m4ac.chan_config = i;
+
+ if (ac->m4ac.chan_config) {
+ int ret = set_default_channel_config(avctx, new_che_pos, ac->m4ac.chan_config);
+ if (!ret)
+ output_configure(ac, ac->che_pos, new_che_pos, ac->m4ac.chan_config, OC_GLOBAL_HDR);
+ else if (avctx->err_recognition & AV_EF_EXPLODE)
+ return AVERROR_INVALIDDATA;
+ }
}
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ output_scale_factor = 1.0 / 32768.0;
+ } else {
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ output_scale_factor = 1.0;
+ }
AAC_INIT_VLC_STATIC( 0, 304);
AAC_INIT_VLC_STATIC( 1, 270);
ff_aac_sbr_init();
dsputil_init(&ac->dsp, avctx);
+ ff_fmt_convert_init(&ac->fmt_conv, avctx);
ac->random_state = 0x1f2e3d4c;
- // -1024 - Compensate wrong IMDCT method.
- // 32768 - Required to scale values to the correct range for the bias method
- // for float to int16 conversion.
-
- 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->sf_offset = 60;
- }
-
ff_aac_tableinit();
INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
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, output_scale_factor/1024.0);
+ ff_mdct_init(&ac->mdct_small, 8, 1, output_scale_factor/128.0);
+ ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0/output_scale_factor);
// 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);
cbrt_tableinit();
+ avcodec_get_frame_defaults(&ac->frame);
+ avctx->coded_frame = &ac->frame;
+
return 0;
}
return 0;
}
+/**
+ * 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;
+
+ ltp->lag = get_bits(gb, 11);
+ ltp->coef = ltp_coef[get_bits(gb, 3)];
+ 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.
*/
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
- GetBitContext *gb, int common_window)
+ GetBitContext *gb)
{
if (get_bits1(gb)) {
av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
- memset(ics, 0, sizeof(IndividualChannelStream));
- return -1;
+ return AVERROR_INVALIDDATA;
}
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = get_bits(gb, 2);
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;
+ return AVERROR_INVALIDDATA;
}
} else if (ac->m4ac.object_type == AOT_AAC_LC) {
av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
- memset(ics, 0, sizeof(IndividualChannelStream));
- return -1;
+ return AVERROR_INVALIDDATA;
} 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);
}
}
}
av_log(ac->avctx, 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 AVERROR_INVALIDDATA;
}
return 0;
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 offset[3] = { global_gain, global_gain - 90, 0 };
+ int clipped_offset;
int noise_flag = 1;
static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
for (g = 0; g < ics->num_window_groups; g++) {
} 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) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "%s (%d) out of range.\n", sf_str[2], offset[2]);
- return -1;
+ clipped_offset = av_clip(offset[2], -155, 100);
+ if (offset[2] != clipped_offset) {
+ av_log_ask_for_sample(ac->avctx, "Intensity stereo "
+ "position clipped (%d -> %d).\nIf you heard an "
+ "audible artifact, there may be a bug in the "
+ "decoder. ", offset[2], clipped_offset);
}
- sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
+ sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
}
} else if (band_type[idx] == NOISE_BT) {
for (; i < run_end; i++, idx++) {
offset[1] += get_bits(gb, 9) - 256;
else
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
- if (offset[1] > 255U) {
- av_log(ac->avctx, AV_LOG_ERROR,
- "%s (%d) out of range.\n", sf_str[1], offset[1]);
- return -1;
+ clipped_offset = av_clip(offset[1], -100, 155);
+ if (offset[1] != clipped_offset) {
+ av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
+ "(%d -> %d).\nIf you heard an audible "
+ "artifact, there may be a bug in the decoder. ",
+ offset[1], clipped_offset);
}
- sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
+ sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
}
} else {
for (; i < run_end; i++, idx++) {
"%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] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
}
}
}
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
unsigned sign, const float *scale)
{
- union float754 s0, s1;
+ union av_intfloat32 s0, s1;
s0.f = s1.f = *scale;
s0.i ^= sign >> 1 << 31;
unsigned sign, const float *scale)
{
unsigned nz = idx >> 12;
- union float754 s = { .f = *scale };
- union float754 t;
+ union av_intfloat32 s = { .f = *scale };
+ union av_intfloat32 t;
- t.i = s.i ^ (sign & 1<<31);
+ t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
- t.i = s.i ^ (sign & 1<<31);
+ t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>2 & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
- t.i = s.i ^ (sign & 1<<31);
+ t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>4 & 3] * t.f;
sign <<= nz & 1; nz >>= 1;
- t.i = s.i ^ (sign & 1<<31);
+ t.i = s.i ^ (sign & 1U<<31);
*dst++ = v[idx>>6 & 3] * t.f;
return dst;
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 8 & 15;
- bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
+ bits = nnz ? GET_CACHE(re, gb) : 0;
LAST_SKIP_BITS(re, gb, nnz);
cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
} while (len -= 4);
GET_VLC(code, re, gb, vlc_tab, 8, 2);
cb_idx = cb_vector_idx[code];
nnz = cb_idx >> 8 & 15;
- sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
+ sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
LAST_SKIP_BITS(re, gb, nnz);
cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
} while (len -= 2);
b += 4;
n = (1 << b) + SHOW_UBITS(re, gb, b);
LAST_SKIP_BITS(re, gb, b);
- *icf++ = cbrt_tab[n] | (bits & 1<<31);
+ *icf++ = cbrt_tab[n] | (bits & 1U<<31);
bits <<= 1;
} else {
unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
- *icf++ = (bits & 1<<31) | v;
+ *icf++ = (bits & 1U<<31) | v;
bits <<= !!v;
}
cb_idx >>= 4;
static av_always_inline float flt16_round(float pf)
{
- union float754 tmp;
+ union av_intfloat32 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;
+ union av_intfloat32 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;
+ union av_intfloat32 pun;
pun.f = pf;
pun.i &= 0xFFFF0000U;
return pun.f;
}
static av_always_inline void predict(PredictorState *ps, float *coef,
- float sf_scale, float inv_sf_scale,
- int output_enable)
+ int output_enable)
{
const float a = 0.953125; // 61.0 / 64
const float alpha = 0.90625; // 29.0 / 32
pv = flt16_round(k1 * r0 + k2 * r1);
if (output_enable)
- *coef += pv * sf_scale;
+ *coef += pv;
- e0 = *coef * inv_sf_scale;
+ e0 = *coef;
e1 = e0 - k1 * r0;
ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
{
int sfb, k;
- float sf_scale = ac->sf_scale, inv_sf_scale = 1 / ac->sf_scale;
if (!sce->ics.predictor_initialized) {
reset_all_predictors(sce->predictor_state);
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(&sce->predictor_state[k], &sce->coeffs[k],
- sf_scale, inv_sf_scale,
sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
}
}
global_gain = get_bits(gb, 8);
if (!common_window && !scale_flag) {
- if (decode_ics_info(ac, ics, gb, 0) < 0)
- return -1;
+ if (decode_ics_info(ac, ics, gb) < 0)
+ return AVERROR_INVALIDDATA;
}
if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
* [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)
+static void apply_intensity_stereo(AACContext *ac, 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 g, group, i, idx = 0;
int c;
float scale;
for (g = 0; g < ics->num_window_groups; g++) {
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];
+ ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
+ coef0 + group * 128 + offsets[i],
+ scale,
+ offsets[i + 1] - offsets[i]);
}
} else {
int bt_run_end = sce1->band_type_run_end[idx];
common_window = get_bits1(gb);
if (common_window) {
- if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
- return -1;
+ if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
+ return AVERROR_INVALIDDATA;
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;
+ if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN))
+ if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
+ decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
ms_present = get_bits(gb, 2);
if (ms_present == 3) {
av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
}
}
- apply_intensity_stereo(cpe, ms_present);
+ apply_intensity_stereo(ac, cpe, ms_present);
return 0;
}
int w, filt, m, i;
int bottom, top, order, start, end, size, inc;
float lpc[TNS_MAX_ORDER];
+ float tmp[TNS_MAX_ORDER];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
}
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[0] = coef[start];
+ for (i = 1; i <= FFMIN(m, order); i++)
+ coef[start] += tmp[i] * lpc[i - 1];
+ for (i = order; i > 0; i--)
+ tmp[i] = tmp[i - 1];
+ }
+ }
}
}
}
+/**
+ * Apply 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;
+
+ if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
+ ac->dsp.vector_fmul(in, in, lwindow_prev, 1024);
+ } else {
+ memset(in, 0, 448 * sizeof(float));
+ ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
+ }
+ if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
+ ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
+ } else {
+ ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
+ memset(in + 1024 + 576, 0, 448 * sizeof(float));
+ }
+ ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
+}
+
+/**
+ * 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 *predTime = sce->ret;
+ float *predFreq = ac->buf_mdct;
+ int16_t num_samples = 2048;
+
+ if (ltp->lag < 1024)
+ num_samples = ltp->lag + 1024;
+ for (i = 0; i < num_samples; i++)
+ predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
+ memset(&predTime[i], 0, (2048 - i) * sizeof(float));
+
+ windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
+
+ if (sce->tns.present)
+ apply_tns(predFreq, &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] += predFreq[i];
+ }
+}
+
+/**
+ * Update the LTP buffer for next frame
+ */
+static void update_ltp(AACContext *ac, SingleChannelElement *sce)
+{
+ IndividualChannelStream *ics = &sce->ics;
+ float *saved = sce->saved;
+ float *saved_ltp = sce->coeffs;
+ 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;
+ int i;
+
+ if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
+ memcpy(saved_ltp, saved, 512 * sizeof(float));
+ memset(saved_ltp + 576, 0, 448 * sizeof(float));
+ ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+ for (i = 0; i < 64; i++)
+ saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
+ } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
+ memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
+ memset(saved_ltp + 576, 0, 448 * sizeof(float));
+ ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
+ for (i = 0; i < 64; i++)
+ saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
+ } else { // LONG_STOP or ONLY_LONG
+ ac->dsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
+ for (i = 0; i < 512; i++)
+ saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
+ }
+
+ memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
+ memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
+}
+
/**
* Conduct IMDCT and windowing.
*/
-static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float bias)
+static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
{
IndividualChannelStream *ics = &sce->ics;
float *in = sce->coeffs;
// imdct
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
for (i = 0; i < 1024; i += 128)
- ff_imdct_half(&ac->mdct_small, buf + i, in + i);
+ ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
} else
- ff_imdct_half(&ac->mdct, buf, in);
+ ac->mdct.imdct_half(&ac->mdct, buf, in);
/* window overlapping
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
*/
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, bias, 512);
+ ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, 512);
} else {
- for (i = 0; i < 448; i++)
- out[i] = saved[i] + bias;
+ memcpy( out, saved, 448 * sizeof(float));
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
- ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, bias, 64);
- ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, bias, 64);
- ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, bias, 64);
- ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, bias, 64);
- ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, bias, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
+ ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
+ ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
} else {
- ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, bias, 64);
- for (i = 576; i < 1024; i++)
- out[i] = buf[i-512] + bias;
+ ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
+ memcpy( out + 576, buf + 64, 448 * sizeof(float));
}
}
// buffer update
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
- for (i = 0; i < 64; i++)
- saved[i] = temp[64 + i] - 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, temp + 64, 64 * sizeof(float));
+ ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
+ ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
+ ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 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));
{
int i;
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;
const int len = 1024 << (ac->m4ac.sbr == 1);
for (i = 0; i < len; i++)
- dest[i] += gain * (src[i] - bias);
+ dest[i] += gain * src[i];
}
/**
static void spectral_to_sample(AACContext *ac)
{
int i, type;
- float imdct_bias = (ac->m4ac.sbr <= 0) ? ac->add_bias : 0.0f;
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 (ac->m4ac.object_type == AOT_AAC_LTP) {
+ if (che->ch[0].ics.predictor_present) {
+ 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)
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], imdct_bias);
+ imdct_and_windowing(ac, &che->ch[0]);
+ if (ac->m4ac.object_type == AOT_AAC_LTP)
+ update_ltp(ac, &che->ch[0]);
if (type == TYPE_CPE) {
- imdct_and_windowing(ac, &che->ch[1], imdct_bias);
+ imdct_and_windowing(ac, &che->ch[1]);
+ if (ac->m4ac.object_type == AOT_AAC_LTP)
+ update_ltp(ac, &che->ch[1]);
}
if (ac->m4ac.sbr > 0) {
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
int size;
AACADTSHeaderInfo hdr_info;
- size = ff_aac_parse_header(gb, &hdr_info);
+ size = avpriv_aac_parse_header(gb, &hdr_info);
if (size > 0) {
- if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
+ if (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->avctx, 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))
+ if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config,
+ FFMAX(ac->output_configured, OC_TRIAL_FRAME)))
return -7;
} else if (ac->output_configured != OC_LOCKED) {
+ ac->m4ac.chan_config = 0;
ac->output_configured = OC_NONE;
}
if (ac->output_configured != OC_LOCKED) {
ac->m4ac.sbr = -1;
ac->m4ac.ps = -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;
}
- 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->avctx->sample_rate)
ac->avctx->sample_rate = hdr_info.sample_rate;
if (hdr_info.num_aac_frames == 1) {
}
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
- int *data_size, GetBitContext *gb)
+ int *got_frame_ptr, GetBitContext *gb)
{
AACContext *ac = avctx->priv_data;
ChannelElement *che = NULL, *che_prev = NULL;
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
- int err, elem_id, data_size_tmp;
- int samples = 0, multiplier;
+ int err, elem_id;
+ int samples = 0, multiplier, audio_found = 0;
if (show_bits(gb, 12) == 0xfff) {
if (parse_adts_frame_header(ac, gb) < 0) {
case TYPE_SCE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
break;
case TYPE_CPE:
err = decode_cpe(ac, gb, che);
+ audio_found = 1;
break;
case TYPE_CCE:
case TYPE_LFE:
err = decode_ics(ac, &che->ch[0], gb, 0, 0);
+ audio_found = 1;
break;
case TYPE_DSE:
avctx->frame_size = samples;
}
- data_size_tmp = samples * avctx->channels * sizeof(int16_t);
- if (*data_size < data_size_tmp) {
- av_log(avctx, 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);
- return -1;
- }
- *data_size = data_size_tmp;
+ if (samples) {
+ /* get output buffer */
+ ac->frame.nb_samples = samples;
+ if ((err = avctx->get_buffer(avctx, &ac->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return err;
+ }
- if (samples)
- ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
+ ac->fmt_conv.float_interleave((float *)ac->frame.data[0],
+ (const float **)ac->output_data,
+ samples, avctx->channels);
+ else
+ ac->fmt_conv.float_to_int16_interleave((int16_t *)ac->frame.data[0],
+ (const float **)ac->output_data,
+ samples, avctx->channels);
- if (ac->output_configured)
+ *(AVFrame *)data = ac->frame;
+ }
+ *got_frame_ptr = !!samples;
+
+ if (ac->output_configured && audio_found)
ac->output_configured = OC_LOCKED;
return 0;
}
static int aac_decode_frame(AVCodecContext *avctx, void *data,
- int *data_size, AVPacket *avpkt)
+ int *got_frame_ptr, AVPacket *avpkt)
{
+ AACContext *ac = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
GetBitContext gb;
int buf_consumed;
int buf_offset;
int err;
+ int new_extradata_size;
+ const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
+ AV_PKT_DATA_NEW_EXTRADATA,
+ &new_extradata_size);
+
+ if (new_extradata) {
+ av_free(avctx->extradata);
+ avctx->extradata = av_mallocz(new_extradata_size +
+ FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!avctx->extradata)
+ return AVERROR(ENOMEM);
+ avctx->extradata_size = new_extradata_size;
+ memcpy(avctx->extradata, new_extradata, new_extradata_size);
+ if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
+ avctx->extradata,
+ avctx->extradata_size*8, 1) < 0)
+ return AVERROR_INVALIDDATA;
+ }
init_get_bits(&gb, buf, buf_size * 8);
- if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
+ if ((err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb)) < 0)
return err;
buf_consumed = (get_bits_count(&gb) + 7) >> 3;
ff_mdct_end(&ac->mdct);
ff_mdct_end(&ac->mdct_small);
+ ff_mdct_end(&ac->mdct_ltp);
return 0;
}
}
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
- GetBitContext *gb)
+ GetBitContext *gb, int asclen)
{
- AVCodecContext *avctx = latmctx->aac_ctx.avctx;
- MPEG4AudioConfig m4ac;
- int config_start_bit = get_bits_count(gb);
- int bits_consumed, esize;
+ AACContext *ac = &latmctx->aac_ctx;
+ AVCodecContext *avctx = ac->avctx;
+ MPEG4AudioConfig m4ac = {0};
+ int config_start_bit = get_bits_count(gb);
+ int sync_extension = 0;
+ int bits_consumed, esize;
+
+ if (asclen) {
+ sync_extension = 1;
+ asclen = FFMIN(asclen, get_bits_left(gb));
+ } else
+ asclen = get_bits_left(gb);
if (config_start_bit % 8) {
av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
"config not byte aligned.\n", 1);
return AVERROR_INVALIDDATA;
- } else {
- bits_consumed =
- decode_audio_specific_config(NULL, avctx, &m4ac,
+ }
+ if (asclen <= 0)
+ return AVERROR_INVALIDDATA;
+ bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
gb->buffer + (config_start_bit / 8),
- get_bits_left(gb) / 8);
+ asclen, sync_extension);
- if (bits_consumed < 0)
- return AVERROR_INVALIDDATA;
+ if (bits_consumed < 0)
+ return AVERROR_INVALIDDATA;
+
+ if (ac->m4ac.sample_rate != m4ac.sample_rate ||
+ ac->m4ac.chan_config != m4ac.chan_config) {
+
+ av_log(avctx, AV_LOG_INFO, "audio config changed\n");
+ latmctx->initialized = 0;
esize = (bits_consumed+7) / 8;
- if (avctx->extradata_size <= esize) {
+ if (avctx->extradata_size < esize) {
av_free(avctx->extradata);
avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata)
avctx->extradata_size = esize;
memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
-
- skip_bits_long(gb, bits_consumed);
}
+ skip_bits_long(gb, bits_consumed);
return bits_consumed;
}
// for all but first stream: use_same_config = get_bits(gb, 1);
if (!audio_mux_version) {
- if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
+ if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
return ret;
} else {
int ascLen = latm_get_value(gb);
- if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
+ if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
return ret;
ascLen -= ret;
skip_bits_long(gb, ascLen);
}
-static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
- AVPacket *avpkt)
+static int latm_decode_frame(AVCodecContext *avctx, void *out,
+ int *got_frame_ptr, AVPacket *avpkt)
{
struct LATMContext *latmctx = avctx->priv_data;
int muxlength, err;
GetBitContext gb;
- if (avpkt->size == 0)
- return 0;
-
init_get_bits(&gb, avpkt->data, avpkt->size * 8);
// check for LOAS sync word
if (!latmctx->initialized) {
if (!avctx->extradata) {
- *out_size = 0;
+ *got_frame_ptr = 0;
return avpkt->size;
} else {
- if ((err = aac_decode_init(avctx)) < 0)
+ if ((err = decode_audio_specific_config(
+ &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.m4ac,
+ avctx->extradata, avctx->extradata_size*8, 1)) < 0)
return err;
latmctx->initialized = 1;
}
return AVERROR_INVALIDDATA;
}
- if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
+ if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb)) < 0)
return err;
return muxlength;
av_cold static int latm_decode_init(AVCodecContext *avctx)
{
struct LATMContext *latmctx = avctx->priv_data;
- int ret;
-
- ret = aac_decode_init(avctx);
+ int ret = aac_decode_init(avctx);
- if (avctx->extradata_size > 0) {
+ if (avctx->extradata_size > 0)
latmctx->initialized = !ret;
- } else {
- latmctx->initialized = 0;
- }
return ret;
}
AVCodec ff_aac_decoder = {
- "aac",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_AAC,
- sizeof(AACContext),
- aac_decode_init,
- NULL,
- aac_decode_close,
- aac_decode_frame,
+ .name = "aac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_AAC,
+ .priv_data_size = sizeof(AACContext),
+ .init = aac_decode_init,
+ .close = aac_decode_close,
+ .decode = aac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
.sample_fmts = (const enum AVSampleFormat[]) {
- AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
};
*/
AVCodec ff_aac_latm_decoder = {
.name = "aac_latm",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AAC_LATM,
.priv_data_size = sizeof(struct LATMContext),
.init = latm_decode_init,
.decode = latm_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
.sample_fmts = (const enum AVSampleFormat[]) {
- AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
.channel_layouts = aac_channel_layout,
};
projects / ffmpeg / blobdiff