* add temporal noise shaping
***********************************/
+#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "put_bits.h"
-#include "dsputil.h"
+#include "internal.h"
#include "mpeg4audio.h"
#include "kbdwin.h"
#include "sinewin.h"
flush_put_bits(&pb);
}
+#define WINDOW_FUNC(type) \
+static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
+ SingleChannelElement *sce, \
+ const float *audio)
+
+WINDOW_FUNC(only_long)
+{
+ 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_buf;
+
+ 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_buf;
+
+ fdsp->vector_fmul(out, audio, lwindow, 1024);
+ memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
+ fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
+ memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
+}
+
+WINDOW_FUNC(long_stop)
+{
+ 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_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);
+ 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_buf;
+ int w;
+
+ for (w = 0; w < 8; w++) {
+ fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
+ out += 128;
+ in += 128;
+ fdsp->vector_fmul_reverse(out, in, swindow, 128);
+ out += 128;
+ }
+}
+
+static void (*const apply_window[4])(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,
+ [LONG_STOP_SEQUENCE] = apply_long_stop_window
+};
+
static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
float *audio)
{
- int i, k;
- 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[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;
- float *output = sce->ret;
-
- if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
- memcpy(output, audio, sizeof(output[0])*1024);
- if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
- memset(output, 0, sizeof(output[0]) * 448);
- for (i = 448; i < 576; i++)
- output[i] = audio[i] * pwindow[i - 448];
- }
- if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
- for (i = 0; i < 1024; i++) {
- output[i+1024] = audio[i + 1024] * lwindow[1024 - i - 1];
- audio[i] = audio[i + 1024] * lwindow[i];
- }
- } else {
- memcpy(output + 1024, audio + 1024, sizeof(output[0]) * 448);
- for (; i < 576; i++)
- output[i+1024] = audio[i+1024] * swindow[576 - i - 1];
- memset(output+1024+576, 0, sizeof(output[0]) * 448);
- memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
- }
+ int i;
+ float *output = sce->ret_buf;
+
+ 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);
- } else {
- for (k = 0; k < 1024; k += 128) {
- for (i = 448 + k; i < 448 + k + 256; i++)
- output[i - 448 - k] = audio[i];
- s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128);
- s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
- s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
- }
- memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
- }
+ else
+ for (i = 0; i < 1024; i += 128)
+ s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
+ memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
}
/**
/**
* 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;
put_bits(&s->pb, 3, TYPE_FIL);
put_bits(&s->pb, 4, FFMIN(namelen, 15));
if (namelen >= 15)
- put_bits(&s->pb, 8, namelen - 16);
+ put_bits(&s->pb, 8, namelen - 14);
put_bits(&s->pb, 4, 0); //extension type - filler
- padbits = 8 - (put_bits_count(&s->pb) & 7);
+ padbits = -put_bits_count(&s->pb) & 7;
avpriv_align_put_bits(&s->pb);
for (i = 0; i < namelen - 2; i++)
put_bits(&s->pb, 8, name[i]);
}
/*
- * 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 float *samples)
+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];
-
- /* deinterleave and remap input samples */
- for (ch = 0; ch < sinc; ch++) {
- const float *sptr = samples + channel_map[ch];
+ int ch;
+ int end = 2048 + (frame ? frame->nb_samples : 0);
+ const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
+ /* 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][0], &s->planar_samples[ch][1024], 1024 * sizeof(s->planar_samples[0][0]));
+ memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
- /* deinterleave */
- for (i = 1024; i < 1024 * 2; i++) {
- s->planar_samples[ch][i] = *sptr;
- sptr += sinc;
+ /* copy new samples and zero any remaining samples */
+ if (frame) {
+ 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][end], 0,
+ (3072 - end) * 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)
+ if (s->last_frame == 2)
return 0;
- if (data) {
- deinterleave_input_samples(s, data);
- if (s->psypp)
- ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
+ /* add current frame to queue */
+ if (frame) {
+ if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
+ return ret;
}
+ copy_input_samples(s, frame);
+ if (s->psypp)
+ ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
+
if (!avctx->frame_number)
return 0;
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;
}
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;
- init_put_bits(&s->pb, frame, buf_size*8);
- if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
- put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
+
+ init_put_bits(&s->pb, avpkt->data, avpkt->size);
+
+ 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) {
avctx->frame_bits = put_bits_count(&s->pb);
// rate control stuff
- if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
+ if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
s->lambda *= ratio;
s->lambda = FFMIN(s->lambda, 65536.f);
}
- if (!data)
- s->last_frame = 1;
+ if (!frame)
+ s->last_frame++;
+
+ ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
+ &avpkt->duration);
- return put_bits_count(&s->pb)>>3;
+ avpkt->size = put_bits_count(&s->pb) >> 3;
+ *got_packet_ptr = 1;
+ return 0;
}
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);
return 0;
}
{
int ret = 0;
- dsputil_init(&s->dsp, avctx);
+ 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);
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
{
+ 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(int ch = 0; ch < s->channels; ch++)
+ for(ch = 0; ch < s->channels; ch++)
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
return 0;
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->initial_padding = 1024;
+ ff_af_queue_init(avctx, &s->afq);
+
return 0;
fail:
aac_encode_end(avctx);
#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",
+ .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,
- .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},
- .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
- .priv_class = &aacenc_class,
+ .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 },
+ .priv_class = &aacenc_class,
};
projects / ffmpeg / blobdiff