xenia-project/FFmpeg
1
0
Fork 0
mirror of https://github.com/xenia-project/FFmpeg.git synced 2024-11-29 22:40:23 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity
d75190aa93
FFmpeg/libavcodec/wmaenc.c
Anton Khirnov 2df0c32ea1 lavc: use a separate field for exporting audio encoder padding 
Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
  meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
  codec context is used for encoding or decoding (and has yet another
  different meaning for video), preventing generic handling of the codec
  context.

Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
2014-10-13 19:09:01 +00:00

454 lines
14 KiB
C
Raw Blame History

/*
* WMA compatible encoder
* Copyright (c) 2007 Michael Niedermayer
*
* This file is part of Libav.
*
* Libav 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,
* 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
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/attributes.h"
#include "avcodec.h"
#include "internal.h"
#include "wma.h"
#undef NDEBUG
#include <assert.h>
static av_cold int encode_init(AVCodecContext *avctx)
{
WMACodecContext *s = avctx->priv_data;
int i, flags1, flags2, block_align;
uint8_t *extradata;
s->avctx = avctx;
if (avctx->channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR,
"too many channels: got %i, need %i or fewer",
avctx->channels, MAX_CHANNELS);
return AVERROR(EINVAL);
}
if (avctx->sample_rate > 48000) {
av_log(avctx, AV_LOG_ERROR, "sample rate is too high: %d > 48kHz",
avctx->sample_rate);
return AVERROR(EINVAL);
}
if (avctx->bit_rate < 24 * 1000) {
av_log(avctx, AV_LOG_ERROR,
"bitrate too low: got %i, need 24000 or higher\n",
avctx->bit_rate);
return AVERROR(EINVAL);
}
/* extract flag infos */
flags1 = 0;
flags2 = 1;
if (avctx->codec->id == AV_CODEC_ID_WMAV1) {
extradata = av_malloc(4);
avctx->extradata_size = 4;
AV_WL16(extradata, flags1);
AV_WL16(extradata + 2, flags2);
} else if (avctx->codec->id == AV_CODEC_ID_WMAV2) {
extradata = av_mallocz(10);
avctx->extradata_size = 10;
AV_WL32(extradata, flags1);
AV_WL16(extradata + 4, flags2);
} else {
assert(0);
}
avctx->extradata = extradata;
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
if (avctx->channels == 2)
s->ms_stereo = 1;
ff_wma_init(avctx, flags2);
/* init MDCT */
for (i = 0; i < s->nb_block_sizes; i++)
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
block_align = avctx->bit_rate * (int64_t) s->frame_len /
(avctx->sample_rate * 8);
block_align = FFMIN(block_align, MAX_CODED_SUPERFRAME_SIZE);
avctx->block_align = block_align;
avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate /
s->frame_len;
avctx->frame_size = avctx->initial_padding = s->frame_len;
return 0;
}
static void apply_window_and_mdct(AVCodecContext *avctx, const AVFrame *frame)
{
WMACodecContext *s = avctx->priv_data;
float **audio = (float **) frame->extended_data;
int len = frame->nb_samples;
int window_index = s->frame_len_bits - s->block_len_bits;
FFTContext *mdct = &s->mdct_ctx[window_index];
int ch;
const float *win = s->windows[window_index];
int window_len = 1 << s->block_len_bits;
float n = 2.0 * 32768.0 / window_len;
for (ch = 0; ch < avctx->channels; ch++) {
memcpy(s->output, s->frame_out[ch], window_len * sizeof(*s->output));
s->fdsp.vector_fmul_scalar(s->frame_out[ch], audio[ch], n, len);
s->fdsp.vector_fmul_reverse(&s->output[window_len], s->frame_out[ch],
win, len);
s->fdsp.vector_fmul(s->frame_out[ch], s->frame_out[ch], win, len);
mdct->mdct_calc(mdct, s->coefs[ch], s->output);
}
}
// FIXME use for decoding too
static void init_exp(WMACodecContext *s, int ch, const int *exp_param)
{
int n;
const uint16_t *ptr;
float v, *q, max_scale, *q_end;
ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
q = s->exponents[ch];
q_end = q + s->block_len;
max_scale = 0;
while (q < q_end) {
/* XXX: use a table */
v = pow(10, *exp_param++ *(1.0 / 16.0));
max_scale = FFMAX(max_scale, v);
n = *ptr++;
do {
*q++ = v;
} while (--n);
}
s->max_exponent[ch] = max_scale;
}
static void encode_exp_vlc(WMACodecContext *s, int ch, const int *exp_param)
{
int last_exp;
const uint16_t *ptr;
float *q, *q_end;
ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
q = s->exponents[ch];
q_end = q + s->block_len;
if (s->version == 1) {
last_exp = *exp_param++;
assert(last_exp - 10 >= 0 && last_exp - 10 < 32);
put_bits(&s->pb, 5, last_exp - 10);
q += *ptr++;
} else
last_exp = 36;
while (q < q_end) {
int exp = *exp_param++;
int code = exp - last_exp + 60;
assert(code >= 0 && code < 120);
put_bits(&s->pb, ff_aac_scalefactor_bits[code],
ff_aac_scalefactor_code[code]);
/* XXX: use a table */
q += *ptr++;
last_exp = exp;
}
}
static int encode_block(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
int total_gain)
{
int v, bsize, ch, coef_nb_bits, parse_exponents;
float mdct_norm;
int nb_coefs[MAX_CHANNELS];
static const int fixed_exp[25] = {
20, 20, 20, 20, 20,
20, 20, 20, 20, 20,
20, 20, 20, 20, 20,
20, 20, 20, 20, 20,
20, 20, 20, 20, 20
};
// FIXME remove duplication relative to decoder
if (s->use_variable_block_len) {
assert(0); // FIXME not implemented
} else {
/* fixed block len */
s->next_block_len_bits = s->frame_len_bits;
s->prev_block_len_bits = s->frame_len_bits;
s->block_len_bits = s->frame_len_bits;
}
s->block_len = 1 << s->block_len_bits;
// assert((s->block_pos + s->block_len) <= s->frame_len);
bsize = s->frame_len_bits - s->block_len_bits;
// FIXME factor
v = s->coefs_end[bsize] - s->coefs_start;
for (ch = 0; ch < s->avctx->channels; ch++)
nb_coefs[ch] = v;
{
int n4 = s->block_len / 2;
mdct_norm = 1.0 / (float) n4;
if (s->version == 1)
mdct_norm *= sqrt(n4);
}
if (s->avctx->channels == 2)
put_bits(&s->pb, 1, !!s->ms_stereo);
for (ch = 0; ch < s->avctx->channels; ch++) {
// FIXME only set channel_coded when needed, instead of always
s->channel_coded[ch] = 1;
if (s->channel_coded[ch])
init_exp(s, ch, fixed_exp);
}
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
WMACoef *coefs1;
float *coefs, *exponents, mult;
int i, n;
coefs1 = s->coefs1[ch];
exponents = s->exponents[ch];
mult = pow(10, total_gain * 0.05) / s->max_exponent[ch];
mult *= mdct_norm;
coefs = src_coefs[ch];
if (s->use_noise_coding && 0) {
assert(0); // FIXME not implemented
} else {
coefs += s->coefs_start;
n = nb_coefs[ch];
for (i = 0; i < n; i++) {
double t = *coefs++ / (exponents[i] * mult);
if (t < -32768 || t > 32767)
return -1;
coefs1[i] = lrint(t);
}
}
}
}
v = 0;
for (ch = 0; ch < s->avctx->channels; ch++) {
int a = s->channel_coded[ch];
put_bits(&s->pb, 1, a);
v |= a;
}
if (!v)
return 1;
for (v = total_gain - 1; v >= 127; v -= 127)
put_bits(&s->pb, 7, 127);
put_bits(&s->pb, 7, v);
coef_nb_bits = ff_wma_total_gain_to_bits(total_gain);
if (s->use_noise_coding) {
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
int i, n;
n = s->exponent_high_sizes[bsize];
for (i = 0; i < n; i++) {
put_bits(&s->pb, 1, s->high_band_coded[ch][i] = 0);
if (0)
nb_coefs[ch] -= s->exponent_high_bands[bsize][i];
}
}
}
}
parse_exponents = 1;
if (s->block_len_bits != s->frame_len_bits)
put_bits(&s->pb, 1, parse_exponents);
if (parse_exponents) {
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
if (s->use_exp_vlc) {
encode_exp_vlc(s, ch, fixed_exp);
} else {
assert(0); // FIXME not implemented
// encode_exp_lsp(s, ch);
}
}
}
} else
assert(0); // FIXME not implemented
for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
int run, tindex;
WMACoef *ptr, *eptr;
tindex = (ch == 1 && s->ms_stereo);
ptr = &s->coefs1[ch][0];
eptr = ptr + nb_coefs[ch];
run = 0;
for (; ptr < eptr; ptr++) {
if (*ptr) {
int level = *ptr;
int abs_level = FFABS(level);
int code = 0;
if (abs_level <= s->coef_vlcs[tindex]->max_level)
if (run < s->coef_vlcs[tindex]->levels[abs_level - 1])
code = run + s->int_table[tindex][abs_level - 1];
assert(code < s->coef_vlcs[tindex]->n);
put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[code],
s->coef_vlcs[tindex]->huffcodes[code]);
if (code == 0) {
if (1 << coef_nb_bits <= abs_level)
return -1;
put_bits(&s->pb, coef_nb_bits, abs_level);
put_bits(&s->pb, s->frame_len_bits, run);
}
// FIXME the sign is flipped somewhere
put_bits(&s->pb, 1, level < 0);
run = 0;
} else
run++;
}
if (run)
put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[1],
s->coef_vlcs[tindex]->huffcodes[1]);
}
if (s->version == 1 && s->avctx->channels >= 2)
avpriv_align_put_bits(&s->pb);
}
return 0;
}
static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
uint8_t *buf, int buf_size, int total_gain)
{
init_put_bits(&s->pb, buf, buf_size);
if (s->use_bit_reservoir)
assert(0); // FIXME not implemented
else if (encode_block(s, src_coefs, total_gain) < 0)
return INT_MAX;
avpriv_align_put_bits(&s->pb);
return put_bits_count(&s->pb) / 8 - s->avctx->block_align;
}
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
WMACodecContext *s = avctx->priv_data;
int i, total_gain, ret;
s->block_len_bits = s->frame_len_bits; // required by non variable block len
s->block_len = 1 << s->block_len_bits;
apply_window_and_mdct(avctx, frame);
if (s->ms_stereo) {
float a, b;
int i;
for (i = 0; i < s->block_len; i++) {
a = s->coefs[0][i] * 0.5;
b = s->coefs[1][i] * 0.5;
s->coefs[0][i] = a + b;
s->coefs[1][i] = a - b;
}
}
if ((ret = ff_alloc_packet(avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
#if 1
total_gain = 128;
for (i = 64; i; i >>= 1) {
int error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
total_gain - i);
if (error < 0)
total_gain -= i;
}
#else
total_gain = 90;
best = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
for (i = 32; i; i >>= 1) {
int scoreL = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain - i);
int scoreR = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain + i);
av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain);
if (scoreL < FFMIN(best, scoreR)) {
best = scoreL;
total_gain -= i;
} else if (scoreR < best) {
best = scoreR;
total_gain += i;
}
}
#endif /* 1 */
if ((i = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain)) >= 0) {
av_log(avctx, AV_LOG_ERROR, "required frame size too large. please "
"use a higher bit rate.\n");
return AVERROR(EINVAL);
}
assert((put_bits_count(&s->pb) & 7) == 0);
while (i++)
put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb);
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
avpkt->size = avctx->block_align;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_wmav1_encoder = {
.name = "wmav1",
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode2 = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
AVCodec ff_wmav2_encoder = {
.name = "wmav2",
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_WMAV2,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode2 = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
Reference in New Issue View Git Blame Copy Permalink