#include "libavutil/opt.h"
#include "avcodec.h"
#include "put_bits.h"
-#include "dsputil.h"
#include "internal.h"
#include "mpeg4audio.h"
#include "kbdwin.h"
32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
};
-static const uint8_t *swb_size_1024[] = {
+static const uint8_t * const swb_size_1024[] = {
swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
};
-static const uint8_t *swb_size_128[] = {
+static const uint8_t * const swb_size_128[] = {
/* the last entry on the following row is swb_size_128_64 but is a
duplicate of swb_size_128_96 */
swb_size_128_96, swb_size_128_96, swb_size_128_96,
}
#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]));
}
}
ChannelElement *cpe;
int i, ch, w, g, chans, tag, start_ch, ret;
int chan_el_counter[4];
+ int frame_bits;
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
if (s->last_frame == 2)
/* 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);
}
do {
- int frame_bits;
-
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);
+ 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));
for (i = 0; i < s->chan_map[0]; i++) {
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;
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) {
put_bits(&s->pb, 3, TYPE_END);
flush_put_bits(&s->pb);
- avctx->frame_bits = put_bits_count(&s->pb);
+ frame_bits = put_bits_count(&s->pb);
+#if FF_API_STAT_BITS
+FF_DISABLE_DEPRECATION_WARNINGS
+ avctx->frame_bits = frame_bits;
+FF_ENABLE_DEPRECATION_WARNINGS
+#endif
// rate control stuff
- if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
- float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
+ if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
+ float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / 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);
ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
"Unsupported profile %d\n", avctx->profile);
ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
- "Too many bits per frame requested\n");
+ "Too many bits %f > %d per frame requested\n",
+ 1024.0 * avctx->bit_rate / avctx->sample_rate,
+ 6144 * s->channels);
s->samplerate_index = i;
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;
#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
static const AVOption aacenc_options[] = {
- {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
- {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.dbl = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
- {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.dbl = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
- {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.dbl = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+ {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
+ {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+ {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
+ {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
{NULL}
};
static const AVClass aacenc_class = {
- "AAC encoder",
- av_default_item_name,
- aacenc_options,
- LIBAVUTIL_VERSION_INT,
+ .class_name = "AAC encoder",
+ .item_name = av_default_item_name,
+ .option = aacenc_options,
+ .version = LIBAVUTIL_VERSION_INT,
};
AVCodec ff_aac_encoder = {
.name = "aac",
+ .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_AAC,
+ .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,
- .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
+ .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("Advanced Audio Coding"),
.priv_class = &aacenc_class,
};