* WMA compatible encoder
* Copyright (c) 2007 Michael Niedermayer
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avcodec.h"
+#include "internal.h"
#include "wma.h"
#undef NDEBUG
static int encode_init(AVCodecContext * avctx){
- WMADecodeContext *s = avctx->priv_data;
- int i, flags1, flags2;
+ WMACodecContext *s = avctx->priv_data;
+ int i, flags1, flags2, block_align;
uint8_t *extradata;
- if(avctx->channels > MAX_CHANNELS)
- return -1;
+ 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 == CODEC_ID_WMAV1) {
+ if (avctx->codec->id == AV_CODEC_ID_WMAV1) {
extradata= av_malloc(4);
avctx->extradata_size= 4;
- extradata[0] = flags1;
- extradata[1] = flags1>>8;
- extradata[2] = flags2;
- extradata[3] = flags2>>8;
- } else if (avctx->codec->id == CODEC_ID_WMAV2) {
+ 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;
- extradata[0] = flags1;
- extradata[1] = flags1>>8;
- extradata[2] = flags1>>16;
- extradata[3] = flags1>>24;
- extradata[4] = flags2;
- extradata[5] = flags2>>8;
+ 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);
-
- avctx->block_align=
- s->block_align= avctx->bit_rate*(int64_t)s->frame_len / (avctx->sample_rate*8);
-//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate);
- avctx->frame_size= s->frame_len;
+ 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->delay = s->frame_len;
+
+#if FF_API_OLD_ENCODE_AUDIO
+ avctx->coded_frame = &s->frame;
+ avcodec_get_frame_defaults(avctx->coded_frame);
+#endif
return 0;
}
-static void apply_window_and_mdct(AVCodecContext * avctx, signed short * audio, int len) {
- WMADecodeContext *s = avctx->priv_data;
+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;
- int i, j, channel;
+ 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 = window_len/2;
-
- for (channel = 0; channel < avctx->channels; channel++) {
- memcpy(s->output, s->frame_out[channel], sizeof(float)*window_len);
- j = channel;
- for (i = 0; i < len; i++, j += avctx->channels){
- s->output[i+window_len] = audio[j] / n * win[i];
- s->frame_out[channel][i] = audio[j] / n * win[window_len - i - 1];
- }
- ff_mdct_calc(&s->mdct_ctx[window_index], s->coefs[channel], s->output, s->mdct_tmp);
+ 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->dsp.vector_fmul_scalar(s->frame_out[ch], audio[ch], n, len);
+ s->dsp.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(WMADecodeContext *s, int ch, int *exp_param){
+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;
s->max_exponent[ch] = max_scale;
}
-static void encode_exp_vlc(WMADecodeContext *s, int ch, const int *exp_param){
+static void encode_exp_vlc(WMACodecContext *s, int ch, const int *exp_param){
int last_exp;
const uint16_t *ptr;
float *q, *q_end;
int exp = *exp_param++;
int code = exp - last_exp + 60;
assert(code >= 0 && code < 120);
- put_bits(&s->pb, ff_wma_scale_huffbits[code], ff_wma_scale_huffcodes[code]);
+ 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(WMADecodeContext *s, float (*src_coefs)[BLOCK_MAX_SIZE], int total_gain){
+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];
//FIXME factor
v = s->coefs_end[bsize] - s->coefs_start;
- for(ch = 0; ch < s->nb_channels; ch++)
+ for (ch = 0; ch < s->avctx->channels; ch++)
nb_coefs[ch] = v;
{
int n4 = s->block_len / 2;
}
}
- if (s->nb_channels == 2) {
- put_bits(&s->pb, 1, s->ms_stereo= 1);
+ if (s->avctx->channels == 2) {
+ put_bits(&s->pb, 1, !!s->ms_stereo);
}
- for(ch = 0; ch < s->nb_channels; ch++) {
- if (s->channel_coded[ch]= 1) { //FIXME
+ for (ch = 0; ch < s->avctx->channels; ch++) {
+ s->channel_coded[ch] = 1; //FIXME only set channel_coded when needed, instead of always
+ if (s->channel_coded[ch]) {
init_exp(s, ch, fixed_exp);
}
}
- for(ch = 0; ch < s->nb_channels; ch++) {
+ for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
- int16_t *coefs1;
+ WMACoef *coefs1;
float *coefs, *exponents, mult;
int i, n;
}
v = 0;
- for(ch = 0; ch < s->nb_channels; ch++) {
+ for (ch = 0; ch < s->avctx->channels; ch++) {
int a = s->channel_coded[ch];
put_bits(&s->pb, 1, a);
v |= a;
coef_nb_bits= ff_wma_total_gain_to_bits(total_gain);
if (s->use_noise_coding) {
- for(ch = 0; ch < s->nb_channels; ch++) {
+ for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
int i, n;
n = s->exponent_high_sizes[bsize];
}
if (parse_exponents) {
- for(ch = 0; ch < s->nb_channels; ch++) {
+ 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);
assert(0); //FIXME not implemented
}
- for(ch = 0; ch < s->nb_channels; ch++) {
+ for (ch = 0; ch < s->avctx->channels; ch++) {
if (s->channel_coded[ch]) {
int run, tindex;
- int16_t *ptr, *eptr;
+ WMACoef *ptr, *eptr;
tindex = (ch == 1 && s->ms_stereo);
ptr = &s->coefs1[ch][0];
eptr = ptr + nb_coefs[ch];
put_bits(&s->pb, coef_nb_bits, abs_level);
put_bits(&s->pb, s->frame_len_bits, run);
}
- put_bits(&s->pb, 1, level > 0);
+ put_bits(&s->pb, 1, level < 0); //FIXME the sign is fliped somewhere
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->nb_channels >= 2) {
- align_put_bits(&s->pb);
+ if (s->version == 1 && s->avctx->channels >= 2) {
+ avpriv_align_put_bits(&s->pb);
}
}
return 0;
}
-static int encode_frame(WMADecodeContext *s, float (*src_coefs)[BLOCK_MAX_SIZE], uint8_t *buf, int buf_size, int total_gain){
+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) {
return INT_MAX;
}
- align_put_bits(&s->pb);
+ avpriv_align_put_bits(&s->pb);
- return put_bits_count(&s->pb)/8 - s->block_align;
+ return put_bits_count(&s->pb) / 8 - s->avctx->block_align;
}
-static int encode_superframe(AVCodecContext *avctx,
- unsigned char *buf, int buf_size, void *data){
- WMADecodeContext *s = avctx->priv_data;
- short *samples = data;
- int i, total_gain, best;
+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, samples, avctx->frame_size);
+ apply_window_and_mdct(avctx, frame);
if (s->ms_stereo) {
float 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, buf, buf_size, total_gain-i);
+ 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, buf, buf_size, total_gain);
+ 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, buf, buf_size, total_gain-i);
- int scoreR= encode_frame(s, s->coefs, buf, buf_size, total_gain+i);
+ 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;
}
#endif
- encode_frame(s, s->coefs, buf, buf_size, total_gain);
+ 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);
- i= s->block_align - (put_bits_count(&s->pb)+7)/8;
- assert(i>=0);
- while(i--)
+ while (i++)
put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb);
- return pbBufPtr(&s->pb) - s->pb.buf;
+
+ if (frame->pts != AV_NOPTS_VALUE)
+ avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+
+ avpkt->size = avctx->block_align;
+ *got_packet_ptr = 1;
+ return 0;
}
-AVCodec wmav1_encoder =
-{
- "wmav1",
- CODEC_TYPE_AUDIO,
- CODEC_ID_WMAV1,
- sizeof(WMADecodeContext),
- encode_init,
- encode_superframe,
- ff_wma_end,
+AVCodec ff_wmav1_encoder = {
+ .name = "wmav1",
+ .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 },
+ .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
};
-AVCodec wmav2_encoder =
-{
- "wmav2",
- CODEC_TYPE_AUDIO,
- CODEC_ID_WMAV2,
- sizeof(WMADecodeContext),
- encode_init,
- encode_superframe,
- ff_wma_end,
+AVCodec ff_wmav2_encoder = {
+ .name = "wmav2",
+ .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 },
+ .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
};
projects / ffmpeg / blobdiff