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aacenc: use AVCodec.encode2()

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pull/4/head
Justin Ruggles 13 years ago
parent 4bf64961a9
commit ad95307f92
3 changed files with 51 additions and 18 deletions
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  1. 3
      libavcodec/Makefile
  2. 63
      libavcodec/aacenc.c
  3. 3
      libavcodec/aacenc.h

3
libavcodec/Makefile
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@ -57,7 +57,8 @@ OBJS-$(CONFIG_AAC_DECODER) += aacdec.o aactab.o aacsbr.o aacps.o \
OBJS-$(CONFIG_AAC_ENCODER) += aacenc.o aaccoder.o \
aacpsy.o aactab.o \
psymodel.o iirfilter.o \
mpeg4audio.o kbdwin.o
mpeg4audio.o kbdwin.o \
audio_frame_queue.o
OBJS-$(CONFIG_AASC_DECODER) += aasc.o msrledec.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc_float.o ac3enc.o ac3tab.o \

63
libavcodec/aacenc.c
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@ -34,6 +34,7 @@
#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
#include "internal.h"
#include "mpeg4audio.h"
#include "kbdwin.h"
#include "sinewin.h"
@ -476,8 +477,7 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
* Deinterleave input samples.
* Channels are reordered from Libav's default order to AAC order.
*/
static void deinterleave_input_samples(AACEncContext *s,
const float *samples, int nb_samples)
static void deinterleave_input_samples(AACEncContext *s, AVFrame *frame)
{
int ch, i;
const int sinc = s->channels;
@ -485,35 +485,43 @@ static void deinterleave_input_samples(AACEncContext *s,
/* 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][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
/* deinterleave */
for (i = 2048; i < 2048 + nb_samples; 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, *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 == 2)
return 0;
deinterleave_input_samples(s, data, data ? avctx->frame_size : 0);
/* 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);
@ -532,7 +540,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
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;
@ -565,7 +573,13 @@ static int aac_encode_frame(AVCodecContext *avctx,
}
do {
int frame_bits;
init_put_bits(&s->pb, frame, buf_size*8);
if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
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;
@ -645,10 +659,15 @@ static int aac_encode_frame(AVCodecContext *avctx,
s->lambda = FFMIN(s->lambda, 65536.f);
}
if (!data)
if (!frame)
s->last_frame++;
return put_bits_count(&s->pb)>>3;
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
&avpkt->duration);
avpkt->size = put_bits_count(&s->pb) >> 3;
*got_packet_ptr = 1;
return 0;
}
static av_cold int aac_encode_end(AVCodecContext *avctx)
@ -662,6 +681,10 @@ 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;
}
@ -695,6 +718,11 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
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);
@ -756,6 +784,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
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);
@ -784,7 +815,7 @@ AVCodec ff_aac_encoder = {
.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},

3
libavcodec/aacenc.h
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@ -27,7 +27,7 @@
#include "dsputil.h"
#include "aac.h"
#include "audio_frame_queue.h"
#include "psymodel.h"
typedef struct AACEncOptions {
@ -71,6 +71,7 @@ typedef struct AACEncContext {
int cur_channel;
int last_frame;
float lambda;
AudioFrameQueue afq;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients

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