*/
/**
- * @file libavcodec/aacenc.c
+ * @file
* AAC encoder
*/
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
return -1;
}
+ if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
+ av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
+ return -1;
+ }
+ if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
+ av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
+ return -1;
+ }
s->samplerate_index = i;
dsputil_init(&s->dsp, avctx);
// window init
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
- ff_sine_window_init(ff_sine_1024, 1024);
- ff_sine_window_init(ff_sine_128, 128);
+ ff_init_ff_sine_windows(10);
+ ff_init_ff_sine_windows(7);
s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
- avctx->extradata = av_malloc(2);
+ avctx->extradata = av_mallocz(2 + FF_INPUT_BUFFER_PADDING_SIZE);
avctx->extradata_size = 2;
put_audio_specific_config(avctx);
lengths[1] = ff_aac_num_swb_128[i];
ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
s->psypp = ff_psy_preprocess_init(avctx);
- s->coder = &ff_aac_coders[0];
+ s->coder = &ff_aac_coders[2];
s->lambda = avctx->global_quality ? avctx->global_quality : 120;
-#if !CONFIG_HARDCODED_TABLES
- for (i = 0; i < 428; i++)
- ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
-#endif /* CONFIG_HARDCODED_TABLES */
+
+ ff_aac_tableinit();
if (avctx->channels > 5)
av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
s->output[i] = sce->saved[i];
}
if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
- j = channel;
- for (i = 0; i < 1024; i++, j += avctx->channels) {
+ for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) {
s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
sce->saved[i] = audio[j] * lwindow[i];
}
} else {
- j = channel;
- for (i = 0; i < 448; i++, j += avctx->channels)
+ for (i = 0, j = channel; i < 448; i++, j += avctx->channels)
s->output[i+1024] = audio[j];
- for (i = 448; i < 576; i++, j += avctx->channels)
+ for (; i < 576; i++, j += avctx->channels)
s->output[i+1024] = audio[j] * swindow[576 - i - 1];
memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
- j = channel;
- for (i = 0; i < 1024; i++, j += avctx->channels)
+ for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
} else {
- j = channel;
for (k = 0; k < 1024; k += 128) {
for (i = 448 + k; i < 448 + k + 256; i++)
s->output[i - 448 - k] = (i < 1024)
s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
}
- j = channel;
- for (i = 0; i < 1024; i++, j += avctx->channels)
+ for (i = 0, j = channel; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
}
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
samples2 = samples + start_ch;
- la = samples2 + 1024 * avctx->channels + start_ch;
+ la = samples2 + (448+64) * avctx->channels + start_ch;
if (!data)
la = NULL;
for (j = 0; j < chans; j++) {
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
for (j = 0; j < chans; j++) {
+ s->cur_channel = start_ch + j;
+ ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
}
cpe->common_window = 0;
}
}
}
+ s->cur_channel = start_ch;
if (cpe->common_window && s->coder->search_for_ms)
s->coder->search_for_ms(s, cpe, s->lambda);
adjust_frame_information(s, cpe, chans);
}
for (j = 0; j < chans; j++) {
s->cur_channel = start_ch + j;
- ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
}
start_ch += chans;
if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
s->lambda *= ratio;
- s->lambda = fminf(s->lambda, 65536.f);
+ s->lambda = FFMIN(s->lambda, 65536.f);
}
if (!data)
AVCodec aac_encoder = {
"aac",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_AAC,
sizeof(AACEncContext),
aac_encode_init,
aac_encode_frame,
aac_encode_end,
- .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
- .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
+ .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
};