* WMA compatible encoder
* Copyright (c) 2007 Michael Niedermayer
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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,
+ * FFmpeg 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
+ * License along with FFmpeg; 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"
+#include "libavutil/avassert.h"
#undef NDEBUG
#include <assert.h>
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);
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;
+ 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;
}
}
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(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);
}
return put_bits_count(&s->pb)/8 - s->block_align;
}
-static int encode_superframe(AVCodecContext *avctx,
- unsigned char *buf, int buf_size, void *data){
+static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
+{
WMACodecContext *s = avctx->priv_data;
- const short *samples = data;
- int i, total_gain;
+ const int16_t *samples = (const int16_t *)frame->data[0];
+ 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, samples, frame->nb_samples);
if (s->ms_stereo) {
float a, b;
}
}
- 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 ((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
- 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);
- while (i++)
+ encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
+ av_assert0((put_bits_count(&s->pb) & 7) == 0);
+ i= s->block_align - (put_bits_count(&s->pb)+7)/8;
+ av_assert0(i>=0);
+ while(i--)
put_bits(&s->pb, 8, 'N');
flush_put_bits(&s->pb);
- return put_bits_ptr(&s->pb) - s->pb.buf;
+ av_assert0(put_bits_ptr(&s->pb) - s->pb.buf == s->block_align);
+
+ if (frame->pts != AV_NOPTS_VALUE)
+ avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+
+ avpkt->size = s->block_align;
+ *got_packet_ptr = 1;
+ return 0;
}
AVCodec ff_wmav1_encoder = {
.id = CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
- .encode = encode_superframe,
+ .encode2 = 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"),
.id = CODEC_ID_WMAV2,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
- .encode = encode_superframe,
+ .encode2 = 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