diff options
Diffstat (limited to 'libavcodec/flacenc.c')
| -rw-r--r-- | libavcodec/flacenc.c | 114 |
1 files changed, 73 insertions, 41 deletions
diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c index 804333d5e9..bc6d00af7f 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 "bswapdsp.h" -#include "get_bits.h" +#include "put_bits.h" #include "golomb.h" #include "internal.h" #include "lpc.h" @@ -80,7 +81,7 @@ typedef struct FlacSubframe { int shift; RiceContext rc; int32_t samples[FLAC_MAX_BLOCKSIZE]; - int32_t residual[FLAC_MAX_BLOCKSIZE+1]; + int32_t residual[FLAC_MAX_BLOCKSIZE+11]; } FlacSubframe; typedef struct FlacFrame { @@ -157,7 +158,7 @@ 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++) { @@ -251,8 +252,11 @@ 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 */ @@ -278,7 +282,8 @@ 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; } @@ -293,7 +298,7 @@ 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]; @@ -332,13 +337,13 @@ 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; } @@ -349,20 +354,20 @@ 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) { @@ -370,7 +375,7 @@ 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); @@ -398,11 +403,33 @@ 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); ff_bswapdsp_init(&s->bdsp); - ff_flacdsp_init(&s->flac_dsp, avctx->sample_fmt, + ff_flacdsp_init(&s->flac_dsp, avctx->sample_fmt, channels, avctx->bits_per_raw_sample); dprint_compression_options(s); @@ -502,6 +529,9 @@ static uint64_t subframe_count_exact(FlacEncodeContext *s, FlacSubframe *sub, /* subframe header */ count += 8; + if (sub->wasted) + count += sub->wasted; + /* subframe */ if (sub->type == FLAC_SUBFRAME_CONSTANT) { count += sub->obits; @@ -583,10 +613,10 @@ static uint64_t calc_optimal_rice_params(RiceContext *rc, int porder, } -static void calc_sums(int pmin, int pmax, uint32_t *data, int n, int pred_order, - uint64_t sums[][MAX_PARTITIONS]) +static void calc_sum_top(int pmax, uint32_t *data, int n, int pred_order, + uint64_t sums[MAX_PARTITIONS]) { - int i, j; + int i; int parts; uint32_t *res, *res_end; @@ -598,17 +628,18 @@ static void calc_sums(int pmin, int pmax, uint32_t *data, int n, int pred_order, uint64_t sum = 0; while (res < res_end) sum += *(res++); - sums[pmax][i] = sum; + sums[i] = sum; res_end += n >> pmax; } - /* sums for lower levels */ - for (i = pmax - 1; i >= pmin; i--) { - parts = (1 << i); - for (j = 0; j < parts; j++) - sums[i][j] = sums[i+1][2*j] + sums[i+1][2*j+1]; - } } +static void calc_sum_next(int level, uint64_t sums[MAX_PARTITIONS]) +{ + int i; + int parts = (1 << level); + for (i = 0; i < parts; i++) + sums[i] = sums[2*i] + sums[2*i+1]; +} static uint64_t calc_rice_params(RiceContext *rc, int pmin, int pmax, int32_t *data, int n, int pred_order) @@ -617,27 +648,30 @@ static uint64_t calc_rice_params(RiceContext *rc, int pmin, int pmax, uint64_t bits[MAX_PARTITION_ORDER+1]; int opt_porder; RiceContext tmp_rc; - uint64_t sums[MAX_PARTITION_ORDER + 1][MAX_PARTITIONS] = { { 0 } }; + uint64_t sums[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; for (i = 0; i < n; i++) rc->udata[i] = (2 * data[i]) ^ (data[i] >> 31); - calc_sums(pmin, pmax, rc->udata, n, pred_order, sums); + calc_sum_top(pmax, rc->udata, n, pred_order, sums); opt_porder = pmin; bits[pmin] = UINT32_MAX; - for (i = pmin; i <= pmax; i++) { - bits[i] = calc_optimal_rice_params(&tmp_rc, i, sums[i], n, pred_order); - if (bits[i] <= bits[opt_porder]) { + for (i = pmax; ; ) { + bits[i] = calc_optimal_rice_params(&tmp_rc, i, sums, n, pred_order); + if (bits[i] < bits[opt_porder]) { opt_porder = i; *rc = tmp_rc; } + if (i == pmin) + break; + calc_sum_next(--i, sums); } return bits[opt_porder]; @@ -1258,10 +1292,8 @@ 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); @@ -1308,7 +1340,7 @@ 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" }, @@ -1343,7 +1375,7 @@ 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 }, |