* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
* add temporal noise shaping
***********************************/
+#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "put_bits.h"
};
/**
- * Table to remap channels from Libav's default order to AAC order.
+ * Table to remap channels from libavcodec's default order to AAC order.
*/
static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
{ 0 },
}
#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);
/*
* Deinterleave input samples.
- * Channels are reordered from Libav's default order to AAC order.
+ * Channels are reordered from libavcodec's default order to AAC order.
*/
static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
{
}
start_ch += chans;
}
+ if ((ret = ff_alloc_packet2(avctx, 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))
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);
if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
goto fail;
s->psypp = ff_psy_preprocess_init(avctx);
- s->coder = &ff_aac_coders[2];
+ s->coder = &ff_aac_coders[s->options.aac_coder];
s->lambda = avctx->global_quality ? avctx->global_quality : 120;
{"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"},
+ {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.dbl = 2}, 0, AAC_CODER_NB-1, AACENC_FLAGS},
{NULL}
};
.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,
+ .supported_samplerates = avpriv_mpeg4audio_sample_rates,
+ .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