* FLAC audio encoder
* Copyright (c) 2006 Justin Ruggles <justin.ruggles@gmail.com>
*
- * 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 "libavutil/md5.h"
#include "avcodec.h"
#include "get_bits.h"
-#include "dsputil.h"
#include "golomb.h"
#include "lpc.h"
#include "flac.h"
FlacFrame frame;
CompressionOptions options;
AVCodecContext *avctx;
- DSPContext dsp;
+ LPCContext lpc_ctx;
struct AVMD5 *md5ctx;
} FlacEncodeContext;
int freq = avctx->sample_rate;
int channels = avctx->channels;
FlacEncodeContext *s = avctx->priv_data;
- int i, level;
+ int i, level, ret;
uint8_t *streaminfo;
s->avctx = avctx;
- dsputil_init(&s->dsp, avctx);
-
- if (avctx->sample_fmt != SAMPLE_FMT_S16)
+ if (avctx->sample_fmt != AV_SAMPLE_FMT_S16)
return -1;
if (channels < 1 || channels > FLAC_MAX_CHANNELS)
s->options.max_partition_order = ((int[]){ 2, 2, 3, 3, 3, 8, 8, 8, 8, 8, 8, 8, 8})[level];
/* set compression option overrides from AVCodecContext */
-#if LIBAVCODEC_VERSION_MAJOR < 53
+#if FF_API_USE_LPC
/* for compatibility with deprecated AVCodecContext.use_lpc */
if (avctx->use_lpc == 0) {
s->options.lpc_type = AV_LPC_TYPE_FIXED;
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
+ ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
+ s->options.max_prediction_order, AV_LPC_TYPE_LEVINSON);
+
dprint_compression_options(s);
- return 0;
+ return ret;
}
}
+static int rice_count_exact(int32_t *res, int n, int k)
+{
+ int i;
+ int count = 0;
+
+ for (i = 0; i < n; i++) {
+ int32_t v = -2 * res[i] - 1;
+ v ^= v >> 31;
+ count += (v >> k) + 1 + k;
+ }
+ return count;
+}
+
+
+static int subframe_count_exact(FlacEncodeContext *s, FlacSubframe *sub,
+ int pred_order)
+{
+ int p, porder, psize;
+ int i, part_end;
+ int count = 0;
+
+ /* subframe header */
+ count += 8;
+
+ /* subframe */
+ if (sub->type == FLAC_SUBFRAME_CONSTANT) {
+ count += sub->obits;
+ } else if (sub->type == FLAC_SUBFRAME_VERBATIM) {
+ count += s->frame.blocksize * sub->obits;
+ } else {
+ /* warm-up samples */
+ count += pred_order * sub->obits;
+
+ /* LPC coefficients */
+ if (sub->type == FLAC_SUBFRAME_LPC)
+ count += 4 + 5 + pred_order * s->options.lpc_coeff_precision;
+
+ /* rice-encoded block */
+ count += 2;
+
+ /* partition order */
+ porder = sub->rc.porder;
+ psize = s->frame.blocksize >> porder;
+ count += 4;
+
+ /* residual */
+ i = pred_order;
+ part_end = psize;
+ for (p = 0; p < 1 << porder; p++) {
+ int k = sub->rc.params[p];
+ count += 4;
+ count += rice_count_exact(&sub->residual[i], part_end - i, k);
+ i = part_end;
+ part_end = FFMIN(s->frame.blocksize, part_end + psize);
+ }
+ }
+
+ return count;
+}
+
+
#define rice_encode_count(sum, n, k) (((n)*((k)+1))+((sum-(n>>1))>>(k)))
/**
if (i == n) {
sub->type = sub->type_code = FLAC_SUBFRAME_CONSTANT;
res[0] = smp[0];
- return sub->obits;
+ return subframe_count_exact(s, sub, 0);
}
/* VERBATIM */
if (frame->verbatim_only || n < 5) {
sub->type = sub->type_code = FLAC_SUBFRAME_VERBATIM;
memcpy(res, smp, n * sizeof(int32_t));
- return sub->obits * n;
+ return subframe_count_exact(s, sub, 0);
}
min_order = s->options.min_prediction_order;
sub->type_code = sub->type | sub->order;
if (sub->order != max_order) {
encode_residual_fixed(res, smp, n, sub->order);
- return find_subframe_rice_params(s, sub, sub->order);
+ find_subframe_rice_params(s, sub, sub->order);
}
- return bits[sub->order];
+ return subframe_count_exact(s, sub, sub->order);
}
/* LPC */
sub->type = FLAC_SUBFRAME_LPC;
- opt_order = ff_lpc_calc_coefs(&s->dsp, smp, n, min_order, max_order,
+ opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, smp, n, min_order, max_order,
s->options.lpc_coeff_precision, coefs, shift, s->options.lpc_type,
s->options.lpc_passes, omethod,
MAX_LPC_SHIFT, 0);
encode_residual_lpc(res, smp, n, sub->order, sub->coefs, sub->shift);
- return find_subframe_rice_params(s, sub, sub->order);
+ find_subframe_rice_params(s, sub, sub->order);
+
+ return subframe_count_exact(s, sub, sub->order);
}
}
-static void output_frame_header(FlacEncodeContext *s)
+static void write_frame_header(FlacEncodeContext *s)
{
FlacFrame *frame;
int crc;
}
-static void output_subframes(FlacEncodeContext *s)
+static void write_subframes(FlacEncodeContext *s)
{
int ch;
}
-static void output_frame_footer(FlacEncodeContext *s)
+static void write_frame_footer(FlacEncodeContext *s)
{
int crc;
flush_put_bits(&s->pb);
static int write_frame(FlacEncodeContext *s, uint8_t *frame, int buf_size)
{
init_put_bits(&s->pb, frame, buf_size);
- output_frame_header(s);
- output_subframes(s);
- output_frame_footer(s);
+ write_frame_header(s);
+ write_subframes(s);
+ write_frame_footer(s);
return put_bits_count(&s->pb) >> 3;
}
{
FlacEncodeContext *s;
const int16_t *samples = data;
- int out_bytes;
+ int frame_bytes, out_bytes;
s = avctx->priv_data;
- if (buf_size < s->max_framesize * 2) {
- av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
- return 0;
- }
-
/* when the last block is reached, update the header in extradata */
if (!data) {
s->max_framesize = s->max_encoded_framesize;
return 0;
}
+ /* change max_framesize for small final frame */
+ if (avctx->frame_size < s->frame.blocksize) {
+ s->max_framesize = ff_flac_get_max_frame_size(avctx->frame_size,
+ s->channels, 16);
+ }
+
init_frame(s);
copy_samples(s, samples);
channel_decorrelation(s);
- encode_frame(s);
-
- out_bytes = write_frame(s, frame, buf_size);
+ frame_bytes = encode_frame(s);
/* fallback to verbatim mode if the compressed frame is larger than it
would be if encoded uncompressed. */
- if (out_bytes > s->max_framesize) {
+ if (frame_bytes > s->max_framesize) {
s->frame.verbatim_only = 1;
- encode_frame(s);
- out_bytes = write_frame(s, frame, buf_size);
+ frame_bytes = encode_frame(s);
}
+ if (buf_size < frame_bytes) {
+ av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
+ return 0;
+ }
+ out_bytes = write_frame(s, frame, buf_size);
+
s->frame_count++;
avctx->coded_frame->pts = s->sample_count;
s->sample_count += avctx->frame_size;
if (avctx->priv_data) {
FlacEncodeContext *s = avctx->priv_data;
av_freep(&s->md5ctx);
+ ff_lpc_end(&s->lpc_ctx);
}
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
}
-AVCodec flac_encoder = {
+AVCodec ff_flac_encoder = {
"flac",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_FLAC,
flac_encode_close,
NULL,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
- .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};