#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
+#include "internal.h"
#include "mpeg4audio.h"
#include "kbdwin.h"
#include "sinewin.h"
flush_put_bits(&pb);
}
+#define WINDOW_FUNC(type) \
+static void apply_ ##type ##_window(DSPContext *dsp, SingleChannelElement *sce, const float *audio)
+
+WINDOW_FUNC(only_long)
+{
+ 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;
+
+ dsp->vector_fmul (out, audio, lwindow, 1024);
+ dsp->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;
+
+ dsp->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);
+ memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
+}
+
+WINDOW_FUNC(long_stop)
+{
+ 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;
+
+ memset(out, 0, sizeof(out[0]) * 448);
+ dsp->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);
+}
+
+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;
+ int w;
+
+ for (w = 0; w < 8; w++) {
+ dsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
+ out += 128;
+ in += 128;
+ dsp->vector_fmul_reverse(out, in, swindow, 128);
+ out += 128;
+ }
+}
+
+static void (*const apply_window[4])(DSPContext *dsp, SingleChannelElement *sce, const float *audio) = {
+ [ONLY_LONG_SEQUENCE] = apply_only_long_window,
+ [LONG_START_SEQUENCE] = apply_long_start_window,
+ [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
+ [LONG_STOP_SEQUENCE] = apply_long_stop_window
+};
+
static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
float *audio)
{
- int i, k;
- 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[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;
+ int i;
float *output = sce->ret;
- if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
- memcpy(output, sce->saved, sizeof(output[0])*1024);
- if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
- memset(output, 0, sizeof(output[0]) * 448);
- for (i = 448; i < 576; i++)
- output[i] = sce->saved[i] * pwindow[i - 448];
- }
- if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
- for (i = 0; i < 1024; i++) {
- output[i+1024] = audio[i] * lwindow[1024 - i - 1];
- sce->saved[i] = audio[i] * lwindow[i];
- }
- } else {
- memcpy(output + 1024, audio, sizeof(output[0]) * 448);
- for (; i < 576; i++)
- output[i+1024] = audio[i] * swindow[576 - i - 1];
- memset(output+1024+576, 0, sizeof(output[0]) * 448);
- memcpy(sce->saved, audio, sizeof(sce->saved[0]) * 1024);
- }
+ apply_window[sce->ics.window_sequence[0]](&s->dsp, sce, audio);
+
+ if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
- } else {
- for (k = 0; k < 1024; k += 128) {
- for (i = 448 + k; i < 448 + k + 256; i++)
- output[i - 448 - k] = (i < 1024)
- ? sce->saved[i]
- : audio[i-1024];
- s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128);
- s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
- s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
- }
- memcpy(sce->saved, audio, sizeof(sce->saved[0]) * 1024);
- }
+ else
+ for (i = 0; i < 1024; i += 128)
+ s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
+ memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
}
/**
put_bits(&s->pb, 3, TYPE_FIL);
put_bits(&s->pb, 4, FFMIN(namelen, 15));
if (namelen >= 15)
- put_bits(&s->pb, 8, namelen - 16);
+ put_bits(&s->pb, 8, namelen - 14);
put_bits(&s->pb, 4, 0); //extension type - filler
- padbits = 8 - (put_bits_count(&s->pb) & 7);
+ padbits = -put_bits_count(&s->pb) & 7;
avpriv_align_put_bits(&s->pb);
for (i = 0; i < namelen - 2; i++)
put_bits(&s->pb, 8, name[i]);
* Deinterleave input samples.
* Channels are reordered from Libav's default order to AAC order.
*/
-static void deinterleave_input_samples(AACEncContext *s,
- const float *samples)
+static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
{
int ch, i;
const int sinc = s->channels;
/* deinterleave and remap input samples */
for (ch = 0; ch < sinc; ch++) {
- const float *sptr = samples + channel_map[ch];
-
/* copy last 1024 samples of previous frame to the start of the current frame */
- memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][1024], 1024 * sizeof(s->planar_samples[0][0]));
+ memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
/* deinterleave */
- for (i = 1024; i < 1024 * 2; i++) {
- s->planar_samples[ch][i] = *sptr;
- sptr += sinc;
+ i = 2048;
+ 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;
+ }
}
+ memset(&s->planar_samples[ch][i], 0,
+ (3072 - i) * sizeof(s->planar_samples[0][0]));
}
}
-static int aac_encode_frame(AVCodecContext *avctx,
- uint8_t *frame, int buf_size, void *data)
+static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
AACEncContext *s = avctx->priv_data;
- float **samples = s->planar_samples, *samples2, *la;
+ float **samples = s->planar_samples, *samples2, *la, *overlap;
ChannelElement *cpe;
- int i, ch, w, g, chans, tag, start_ch;
+ int i, ch, w, g, chans, tag, start_ch, ret;
int chan_el_counter[4];
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
- if (s->last_frame)
+ if (s->last_frame == 2)
return 0;
- if (data) {
- deinterleave_input_samples(s, data);
- if (s->psypp)
- ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
+ /* add current frame to queue */
+ if (frame) {
+ if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
+ return ret;
}
+ deinterleave_input_samples(s, frame);
+ if (s->psypp)
+ ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
+
if (!avctx->frame_number)
return 0;
for (ch = 0; ch < chans; ch++) {
IndividualChannelStream *ics = &cpe->ch[ch].ics;
int cur_channel = start_ch + ch;
- samples2 = &samples[cur_channel][0];
+ overlap = &samples[cur_channel][0];
+ samples2 = overlap + 1024;
la = samples2 + (448+64);
- if (!data)
+ if (!frame)
la = NULL;
if (tag == TYPE_LFE) {
wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
for (w = 0; w < ics->num_windows; w++)
ics->group_len[w] = wi[ch].grouping[w];
- apply_window_and_mdct(s, &cpe->ch[ch], samples2);
+ apply_window_and_mdct(s, &cpe->ch[ch], overlap);
}
start_ch += chans;
}
+ if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
+ }
+
do {
int frame_bits;
- init_put_bits(&s->pb, frame, buf_size*8);
+
+ 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);
start_ch = 0;
s->lambda = FFMIN(s->lambda, 65536.f);
}
- if (!data)
- s->last_frame = 1;
+ if (!frame)
+ s->last_frame++;
+
+ ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
+ &avpkt->duration);
- return put_bits_count(&s->pb)>>3;
+ avpkt->size = put_bits_count(&s->pb) >> 3;
+ *got_packet_ptr = 1;
+ return 0;
}
static av_cold int aac_encode_end(AVCodecContext *avctx)
ff_psy_preprocess_end(s->psypp);
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;
- dsputil_init(&s->dsp, avctx);
+ ff_dsputil_init(&s->dsp, avctx);
// window init
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
{
- FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 2 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
+ 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);
- for(int ch = 0; ch < s->channels; ch++)
- s->planar_samples[ch] = s->buffer.samples + 2 * 1024 * ch;
+ 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:
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;
+ ff_af_queue_init(avctx, &s->afq);
+
return 0;
fail:
aac_encode_end(avctx);
.id = CODEC_ID_AAC,
.priv_data_size = sizeof(AACEncContext),
.init = aac_encode_init,
- .encode = aac_encode_frame,
+ .encode2 = aac_encode_frame,
.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,AV_SAMPLE_FMT_NONE},
- .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
- .priv_class = &aacenc_class,
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
+ CODEC_CAP_EXPERIMENTAL,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_NONE },
+ .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
+ .priv_class = &aacenc_class,
};
projects / ffmpeg / blobdiff