* add temporal noise shaping
***********************************/
+#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "put_bits.h"
}
#define WINDOW_FUNC(type) \
-static void apply_ ##type ##_window(DSPContext *dsp, SingleChannelElement *sce, const float *audio)
+static void apply_ ##type ##_window(DSPContext *dsp, AVFloatDSPContext *fdsp, \
+ SingleChannelElement *sce, \
+ const float *audio)
WINDOW_FUNC(only_long)
{
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);
+ fdsp->vector_fmul (out, audio, lwindow, 1024);
dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 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);
+ 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);
memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
float *out = sce->ret;
memset(out, 0, sizeof(out[0]) * 448);
- dsp->vector_fmul(out + 448, audio + 448, swindow, 128);
+ 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);
}
int w;
for (w = 0; w < 8; w++) {
- dsp->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);
}
}
-static void (*const apply_window[4])(DSPContext *dsp, SingleChannelElement *sce, const float *audio) = {
+static void (*const apply_window[4])(DSPContext *dsp, AVFloatDSPContext *fdsp,
+ 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,
int i;
float *output = sce->ret;
- apply_window[sce->ics.window_sequence[0]](&s->dsp, sce, audio);
+ apply_window[sce->ics.window_sequence[0]](&s->dsp, &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]));
}
}
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);
}
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;
- 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);
+ 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) {
int ret = 0;
ff_dsputil_init(&s->dsp, avctx);
+ avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
// window init
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
#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}
};
AVCodec ff_aac_encoder = {
.name = "aac",
.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,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_FLTP,
+ AV_SAMPLE_FMT_NONE },
+ .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
+ .priv_class = &aacenc_class,
};
projects / ffmpeg / blobdiff