#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);
/**
* Produce integer coefficients from scalefactors provided by the model.
*/
-static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
+static void adjust_frame_information(ChannelElement *cpe, int chans)
{
int i, w, w2, g, ch;
int start, maxsfb, cmaxsfb;
/**
* Write some auxiliary information about the created AAC file.
*/
-static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
- const char *name)
+static void put_bitstream_info(AACEncContext *s, const char *name)
{
int i, namelen, padbits;
}
/*
- * Deinterleave input samples.
+ * Copy input samples.
* Channels are reordered from Libav's default order to AAC order.
*/
-static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
+static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
{
- int ch, i;
- const int sinc = s->channels;
- const uint8_t *channel_map = aac_chan_maps[sinc - 1];
+ int ch;
+ int end = 2048 + (frame ? frame->nb_samples : 0);
+ const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
- /* deinterleave and remap input samples */
- for (ch = 0; ch < sinc; ch++) {
+ /* copy and remap input samples */
+ for (ch = 0; ch < s->channels; ch++) {
/* copy last 1024 samples of previous frame to the start of the current frame */
memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
- /* deinterleave */
- i = 2048;
+ /* copy new samples and zero any remaining samples */
if (frame) {
- const float *sptr = ((const float *)frame->data[0]) + channel_map[ch];
- for (; i < 2048 + frame->nb_samples; i++) {
- s->planar_samples[ch][i] = *sptr;
- sptr += sinc;
- }
+ memcpy(&s->planar_samples[ch][2048],
+ frame->extended_data[channel_map[ch]],
+ frame->nb_samples * sizeof(s->planar_samples[0][0]));
}
- memset(&s->planar_samples[ch][i], 0,
- (3072 - i) * sizeof(s->planar_samples[0][0]));
+ memset(&s->planar_samples[ch][end], 0,
+ (3072 - end) * sizeof(s->planar_samples[0][0]));
}
}
/* 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;
}
- deinterleave_input_samples(s, frame);
+ copy_input_samples(s, frame);
if (s->psypp)
ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
init_put_bits(&s->pb, avpkt->data, avpkt->size);
if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
- put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
+ put_bitstream_info(s, LIBAVCODEC_IDENT);
start_ch = 0;
memset(chan_el_counter, 0, sizeof(chan_el_counter));
for (i = 0; i < s->chan_map[0]; i++) {
s->coder->search_for_ms(s, cpe, s->lambda);
}
}
- adjust_frame_information(s, cpe, chans);
+ adjust_frame_information(cpe, chans);
if (chans == 2) {
put_bits(&s->pb, 1, cpe->common_window);
if (cpe->common_window) {
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);
// window init
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);
.close = aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
CODEC_CAP_EXPERIMENTAL,
- .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
+ .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,