* 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
*/
s->avctx = avctx;
- if(avctx->channels > MAX_CHANNELS)
- return -1;
+ 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)
- return -1;
+ 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;
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);
+ ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
+
+ s->block_align = avctx->bit_rate * (int64_t)s->frame_len /
+ (avctx->sample_rate * 8);
+ s->block_align = FFMIN(s->block_align, MAX_CODED_SUPERFRAME_SIZE);
+ avctx->block_align = s->block_align;
+ avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate /
+ s->frame_len;
//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;
}
-static void apply_window_and_mdct(AVCodecContext * avctx, signed short * audio, int len) {
+static void apply_window_and_mdct(AVCodecContext * avctx, const signed short * audio, int len) {
WMACodecContext *s = avctx->priv_data;
int window_index= s->frame_len_bits - s->block_len_bits;
+ FFTContext *mdct = &s->mdct_ctx[window_index];
int i, j, channel;
const float * win = s->windows[window_index];
int window_len = 1 << s->block_len_bits;
s->output[i+window_len] = audio[j] / n * win[window_len - i - 1];
s->frame_out[channel][i] = audio[j] / n * win[i];
}
- ff_mdct_calc(&s->mdct_ctx[window_index], s->coefs[channel], s->output, s->mdct_tmp);
+ mdct->mdct_calc(mdct, s->coefs[channel], s->output);
}
}
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;
}
if (s->nb_channels == 2) {
- put_bits(&s->pb, 1, s->ms_stereo= 1);
+ put_bits(&s->pb, 1, !!s->ms_stereo);
}
for(ch = 0; ch < s->nb_channels; ch++) {
- if ((s->channel_coded[ch]= 1)) { //FIXME only set channel_coded when needed, instead of always
+ 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++) {
if (s->channel_coded[ch]) {
- int16_t *coefs1;
+ WMACoef *coefs1;
float *coefs, *exponents, mult;
int i, n;
for(ch = 0; ch < s->nb_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];
if(1<<coef_nb_bits <= abs_level)
return -1;
+
+ //Workaround minor rounding differences for the regression tests, FIXME we should find and replace the problematic float by fixpoint for reg tests
+ if(abs_level == 0x71B && (s->avctx->flags & CODEC_FLAG_BITEXACT)) abs_level=0x71A;
+
put_bits(&s->pb, coef_nb_bits, abs_level);
put_bits(&s->pb, s->frame_len_bits, 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);
+ avpriv_align_put_bits(&s->pb);
}
}
return 0;
return INT_MAX;
}
- align_put_bits(&s->pb);
+ avpriv_align_put_bits(&s->pb);
return put_bits_count(&s->pb)/8 - s->block_align;
}
static int encode_superframe(AVCodecContext *avctx,
unsigned char *buf, int buf_size, void *data){
WMACodecContext *s = avctx->priv_data;
- short *samples = data;
+ const short *samples = data;
int i, total_gain;
s->block_len_bits= s->frame_len_bits; //required by non variable block len
}
}
+ if (buf_size < 2 * MAX_CODED_SUPERFRAME_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "output buffer size is too small\n");
+ return AVERROR(EINVAL);
+ }
+
#if 1
total_gain= 128;
for(i=64; i; i>>=1){
}
#endif
- encode_frame(s, s->coefs, buf, buf_size, total_gain);
+ if ((i = encode_frame(s, s->coefs, buf, buf_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;
+ return s->block_align;
}
-AVCodec wmav1_encoder =
-{
- "wmav1",
- CODEC_TYPE_AUDIO,
- CODEC_ID_WMAV1,
- sizeof(WMACodecContext),
- encode_init,
- encode_superframe,
- ff_wma_end,
- .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
+AVCodec ff_wmav1_encoder = {
+ .name = "wmav1",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_WMAV1,
+ .priv_data_size = sizeof(WMACodecContext),
+ .init = encode_init,
+ .encode = encode_superframe,
+ .close = ff_wma_end,
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,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(WMACodecContext),
- encode_init,
- encode_superframe,
- ff_wma_end,
- .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
+AVCodec ff_wmav2_encoder = {
+ .name = "wmav2",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_WMAV2,
+ .priv_data_size = sizeof(WMACodecContext),
+ .init = encode_init,
+ .encode = encode_superframe,
+ .close = ff_wma_end,
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+ .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
};
projects / ffmpeg / blobdiff