#include "libavutil/opt.h"
#include "avcodec.h"
#include "put_bits.h"
-#include "dsputil.h"
#include "internal.h"
#include "mpeg4audio.h"
#include "kbdwin.h"
}
#define WINDOW_FUNC(type) \
-static void apply_ ##type ##_window(DSPContext *dsp, AVFloatDSPContext *fdsp, \
+static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
SingleChannelElement *sce, \
const float *audio)
{
const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
- float *out = sce->ret;
+ float *out = sce->ret_buf;
- fdsp->vector_fmul (out, audio, lwindow, 1024);
- dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
+ fdsp->vector_fmul (out, audio, lwindow, 1024);
+ fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
}
WINDOW_FUNC(long_start)
{
const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
- float *out = sce->ret;
+ float *out = sce->ret_buf;
fdsp->vector_fmul(out, audio, lwindow, 1024);
memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
- dsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
+ fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
}
{
const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
- float *out = sce->ret;
+ float *out = sce->ret_buf;
memset(out, 0, sizeof(out[0]) * 448);
fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
- dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
+ fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
}
WINDOW_FUNC(eight_short)
const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
const float *in = audio + 448;
- float *out = sce->ret;
+ float *out = sce->ret_buf;
int w;
for (w = 0; w < 8; w++) {
- fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
+ fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
out += 128;
in += 128;
- dsp->vector_fmul_reverse(out, in, swindow, 128);
+ fdsp->vector_fmul_reverse(out, in, swindow, 128);
out += 128;
}
}
-static void (*const apply_window[4])(DSPContext *dsp, AVFloatDSPContext *fdsp,
+static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
SingleChannelElement *sce,
const float *audio) = {
[ONLY_LONG_SEQUENCE] = apply_only_long_window,
float *audio)
{
int i;
- float *output = sce->ret;
+ float *output = sce->ret_buf;
- apply_window[sce->ics.window_sequence[0]](&s->dsp, &s->fdsp, sce, audio);
+ apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
/* add current frame to queue */
if (frame) {
- if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
+ if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
return ret;
}
init_put_bits(&s->pb, avpkt->data, avpkt->size);
- if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
+ if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
put_bitstream_info(s, LIBAVCODEC_IDENT);
start_ch = 0;
memset(chan_el_counter, 0, sizeof(chan_el_counter));
coeffs[ch] = cpe->ch[ch].coeffs;
s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
for (ch = 0; ch < chans; ch++) {
- s->cur_channel = start_ch * 2 + ch;
+ s->cur_channel = start_ch + ch;
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
}
cpe->common_window = 0;
}
}
}
- s->cur_channel = start_ch * 2;
+ s->cur_channel = start_ch;
if (s->options.stereo_mode && cpe->common_window) {
if (s->options.stereo_mode > 0) {
IndividualChannelStream *ics = &cpe->ch[0].ics;
avctx->frame_bits = put_bits_count(&s->pb);
// rate control stuff
- if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
+ if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
s->lambda *= ratio;
s->lambda = FFMIN(s->lambda, 65536.f);
av_freep(&s->buffer.samples);
av_freep(&s->cpe);
ff_af_queue_close(&s->afq);
-#if FF_API_OLD_ENCODE_AUDIO
- av_freep(&avctx->coded_frame);
-#endif
return 0;
}
{
int ret = 0;
- ff_dsputil_init(&s->dsp, avctx);
- avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+ avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
// window init
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
int ch;
FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
- FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
+ FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
for(ch = 0; ch < s->channels; ch++)
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
-#if FF_API_OLD_ENCODE_AUDIO
- if (!(avctx->coded_frame = avcodec_alloc_frame()))
- goto alloc_fail;
-#endif
-
return 0;
alloc_fail:
return AVERROR(ENOMEM);
s->chan_map = aac_chan_configs[s->channels-1];
- if (ret = dsp_init(avctx, s))
+ if ((ret = dsp_init(avctx, s)) < 0)
goto fail;
- if (ret = alloc_buffers(avctx, s))
+ if ((ret = alloc_buffers(avctx, s)) < 0)
goto fail;
avctx->extradata_size = 5;
lengths[1] = ff_aac_num_swb_128[i];
for (i = 0; i < s->chan_map[0]; i++)
grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
- if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
+ if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
+ s->chan_map[0], grouping)) < 0)
goto fail;
s->psypp = ff_psy_preprocess_init(avctx);
s->coder = &ff_aac_coders[2];
for (i = 0; i < 428; i++)
ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
- avctx->delay = 1024;
+ avctx->initial_padding = 1024;
ff_af_queue_init(avctx, &s->afq);
return 0;
AVCodec ff_aac_encoder = {
.name = "aac",
+ .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AAC,
.priv_data_size = sizeof(AACEncContext),
.init = aac_encode_init,
.encode2 = aac_encode_frame,
.close = aac_encode_end,
- .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
- CODEC_CAP_EXPERIMENTAL,
+ .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
+ AV_CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
- .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.priv_class = &aacenc_class,
};