summaryrefslogtreecommitdiff
path: root/libavcodec/flacenc.c
diff options
context:
space:
mode:
Diffstat (limited to 'libavcodec/flacenc.c')
-rw-r--r--libavcodec/flacenc.c73
1 files changed, 49 insertions, 24 deletions
diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c
index 58e86be2d6..1fc8c4ce6b 100644
--- a/libavcodec/flacenc.c
+++ b/libavcodec/flacenc.c
@@ -2,30 +2,31 @@
* FLAC audio encoder
* Copyright (c) 2006 Justin Ruggles <justin.ruggles@gmail.com>
*
- * 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 "libavutil/avassert.h"
#include "libavutil/crc.h"
#include "libavutil/intmath.h"
#include "libavutil/md5.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "dsputil.h"
-#include "get_bits.h"
+#include "put_bits.h"
#include "golomb.h"
#include "internal.h"
#include "lpc.h"
@@ -153,7 +154,7 @@ static int select_blocksize(int samplerate, int block_time_ms)
int target;
int blocksize;
- assert(samplerate > 0);
+ av_assert0(samplerate > 0);
blocksize = ff_flac_blocksize_table[1];
target = (samplerate * block_time_ms) / 1000;
for (i = 0; i < 16; i++) {
@@ -247,8 +248,11 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
break;
}
- if (channels < 1 || channels > FLAC_MAX_CHANNELS)
- return -1;
+ if (channels < 1 || channels > FLAC_MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "%d channels not supported (max %d)\n",
+ channels, FLAC_MAX_CHANNELS);
+ return AVERROR(EINVAL);
+ }
s->channels = channels;
/* find samplerate in table */
@@ -274,7 +278,8 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
s->sr_code[0] = 13;
s->sr_code[1] = freq;
} else {
- return -1;
+ av_log(avctx, AV_LOG_ERROR, "%d Hz not supported\n", freq);
+ return AVERROR(EINVAL);
}
s->samplerate = freq;
}
@@ -289,7 +294,7 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
if (level > 12) {
av_log(avctx, AV_LOG_ERROR, "invalid compression level: %d\n",
s->options.compression_level);
- return -1;
+ return AVERROR(EINVAL);
}
s->options.block_time_ms = ((int[]){ 27, 27, 27,105,105,105,105,105,105,105,105,105,105})[level];
@@ -328,13 +333,13 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
if (avctx->min_prediction_order > MAX_FIXED_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n",
avctx->min_prediction_order);
- return -1;
+ return AVERROR(EINVAL);
}
} else if (avctx->min_prediction_order < MIN_LPC_ORDER ||
avctx->min_prediction_order > MAX_LPC_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n",
avctx->min_prediction_order);
- return -1;
+ return AVERROR(EINVAL);
}
s->options.min_prediction_order = avctx->min_prediction_order;
}
@@ -345,20 +350,20 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
if (avctx->max_prediction_order > MAX_FIXED_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n",
avctx->max_prediction_order);
- return -1;
+ return AVERROR(EINVAL);
}
} else if (avctx->max_prediction_order < MIN_LPC_ORDER ||
avctx->max_prediction_order > MAX_LPC_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n",
avctx->max_prediction_order);
- return -1;
+ return AVERROR(EINVAL);
}
s->options.max_prediction_order = avctx->max_prediction_order;
}
if (s->options.max_prediction_order < s->options.min_prediction_order) {
av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
s->options.min_prediction_order, s->options.max_prediction_order);
- return -1;
+ return AVERROR(EINVAL);
}
if (avctx->frame_size > 0) {
@@ -366,7 +371,7 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
avctx->frame_size > FLAC_MAX_BLOCKSIZE) {
av_log(avctx, AV_LOG_ERROR, "invalid block size: %d\n",
avctx->frame_size);
- return -1;
+ return AVERROR(EINVAL);
}
} else {
s->avctx->frame_size = select_blocksize(s->samplerate, s->options.block_time_ms);
@@ -394,6 +399,28 @@ static av_cold int flac_encode_init(AVCodecContext *avctx)
s->frame_count = 0;
s->min_framesize = s->max_framesize;
+ if (channels == 3 &&
+ avctx->channel_layout != (AV_CH_LAYOUT_STEREO|AV_CH_FRONT_CENTER) ||
+ channels == 4 &&
+ avctx->channel_layout != AV_CH_LAYOUT_2_2 &&
+ avctx->channel_layout != AV_CH_LAYOUT_QUAD ||
+ channels == 5 &&
+ avctx->channel_layout != AV_CH_LAYOUT_5POINT0 &&
+ avctx->channel_layout != AV_CH_LAYOUT_5POINT0_BACK ||
+ channels == 6 &&
+ avctx->channel_layout != AV_CH_LAYOUT_5POINT1 &&
+ avctx->channel_layout != AV_CH_LAYOUT_5POINT1_BACK) {
+ if (avctx->channel_layout) {
+ av_log(avctx, AV_LOG_ERROR, "Channel layout not supported by Flac, "
+ "output stream will have incorrect "
+ "channel layout.\n");
+ } else {
+ av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The encoder "
+ "will use Flac channel layout for "
+ "%d channels.\n", channels);
+ }
+ }
+
ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
s->options.max_prediction_order, FF_LPC_TYPE_LEVINSON);
@@ -616,9 +643,9 @@ static uint64_t calc_rice_params(RiceContext *rc, int pmin, int pmax,
uint32_t *udata;
uint64_t sums[MAX_PARTITION_ORDER+1][MAX_PARTITIONS];
- assert(pmin >= 0 && pmin <= MAX_PARTITION_ORDER);
- assert(pmax >= 0 && pmax <= MAX_PARTITION_ORDER);
- assert(pmin <= pmax);
+ av_assert1(pmin >= 0 && pmin <= MAX_PARTITION_ORDER);
+ av_assert1(pmax >= 0 && pmax <= MAX_PARTITION_ORDER);
+ av_assert1(pmin <= pmax);
tmp_rc.coding_mode = rc->coding_mode;
@@ -1243,10 +1270,8 @@ static int flac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
}
- if ((ret = ff_alloc_packet(avpkt, frame_bytes))) {
- av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ if ((ret = ff_alloc_packet2(avctx, avpkt, frame_bytes)) < 0)
return ret;
- }
out_bytes = write_frame(s, avpkt);
@@ -1290,7 +1315,7 @@ static const AVOption options[] = {
{ "fixed", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = FF_LPC_TYPE_FIXED }, INT_MIN, INT_MAX, FLAGS, "lpc_type" },
{ "levinson", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = FF_LPC_TYPE_LEVINSON }, INT_MIN, INT_MAX, FLAGS, "lpc_type" },
{ "cholesky", NULL, 0, AV_OPT_TYPE_CONST, {.i64 = FF_LPC_TYPE_CHOLESKY }, INT_MIN, INT_MAX, FLAGS, "lpc_type" },
-{ "lpc_passes", "Number of passes to use for Cholesky factorization during LPC analysis", offsetof(FlacEncodeContext, options.lpc_passes), AV_OPT_TYPE_INT, {.i64 = 1 }, 1, INT_MAX, FLAGS },
+{ "lpc_passes", "Number of passes to use for Cholesky factorization during LPC analysis", offsetof(FlacEncodeContext, options.lpc_passes), AV_OPT_TYPE_INT, {.i64 = 2 }, 1, INT_MAX, FLAGS },
{ "min_partition_order", NULL, offsetof(FlacEncodeContext, options.min_partition_order), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, MAX_PARTITION_ORDER, FLAGS },
{ "max_partition_order", NULL, offsetof(FlacEncodeContext, options.max_partition_order), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, MAX_PARTITION_ORDER, FLAGS },
{ "prediction_order_method", "Search method for selecting prediction order", offsetof(FlacEncodeContext, options.prediction_order_method), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, ORDER_METHOD_LOG, FLAGS, "predm" },
@@ -1325,7 +1350,7 @@ AVCodec ff_flac_encoder = {
.init = flac_encode_init,
.encode2 = flac_encode_frame,
.close = flac_encode_close,
- .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
+ .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_LOSSLESS,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_NONE },