* FLAC (Free Lossless Audio Codec) decoder
* Copyright (c) 2003 Alex Beregszaszi
*
- * 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
*/
FLACSTREAMINFO
AVCodecContext *avctx; ///< parent AVCodecContext
+ AVFrame frame;
GetBitContext gb; ///< GetBitContext initialized to start at the current frame
int blocksize; ///< number of samples in the current frame
allocate_buffers(s);
s->got_streaminfo = 1;
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
}
return 0;
}
-static int flac_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int flac_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
FLACContext *s = avctx->priv_data;
int i, j = 0, bytes_read = 0;
- int16_t *samples_16 = data;
- int32_t *samples_32 = data;
- int alloc_data_size= *data_size;
- int output_size;
+ int16_t *samples_16;
+ int32_t *samples_32;
+ int ret;
- *data_size=0;
+ *got_frame_ptr = 0;
if (s->max_framesize == 0) {
s->max_framesize =
}
bytes_read = (get_bits_count(&s->gb)+7)/8;
- /* check if allocated data size is large enough for output */
- output_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
- if (output_size > alloc_data_size) {
- av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
- "allocated data size\n");
- return -1;
+ /* get output buffer */
+ s->frame.nb_samples = s->blocksize;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
}
- *data_size = output_size;
+ samples_16 = (int16_t *)s->frame.data[0];
+ samples_32 = (int32_t *)s->frame.data[0];
#define DECORRELATE(left, right)\
assert(s->channels == 2);\
buf_size - bytes_read, buf_size);
}
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
+
return bytes_read;
}
.init = flac_decode_init,
.close = flac_decode_close,
.decode = flac_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};