diff options
Diffstat (limited to 'libavcodec/flacdec.c')
| -rw-r--r-- | libavcodec/flacdec.c | 210 |
1 files changed, 157 insertions, 53 deletions
diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c index 6cce692..30fe416 100644 --- a/libavcodec/flacdec.c +++ b/libavcodec/flacdec.c @@ -2,20 +2,20 @@ * 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 */ @@ -33,6 +33,9 @@ #include <limits.h> +#include "libavutil/avassert.h" +#include "libavutil/crc.h" +#include "libavutil/opt.h" #include "avcodec.h" #include "internal.h" #include "get_bits.h" @@ -41,9 +44,13 @@ #include "flac.h" #include "flacdata.h" #include "flacdsp.h" +#include "thread.h" +#include "unary.h" + typedef struct FLACContext { - FLACSTREAMINFO + AVClass *class; + struct FLACStreaminfo flac_stream_info; AVCodecContext *avctx; ///< parent AVCodecContext GetBitContext gb; ///< GetBitContext initialized to start at the current frame @@ -56,6 +63,7 @@ typedef struct FLACContext { int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples uint8_t *decoded_buffer; unsigned int decoded_buffer_size; + int buggy_lpc; ///< use workaround for old lavc encoded files FLACDSPContext dsp; } FLACContext; @@ -65,7 +73,7 @@ static int allocate_buffers(FLACContext *s); static void flac_set_bps(FLACContext *s) { enum AVSampleFormat req = s->avctx->request_sample_fmt; - int need32 = s->bps > 16; + int need32 = s->flac_stream_info.bps > 16; int want32 = av_get_bytes_per_sample(req) > 2; int planar = av_sample_fmt_is_planar(req); @@ -74,13 +82,13 @@ static void flac_set_bps(FLACContext *s) s->avctx->sample_fmt = AV_SAMPLE_FMT_S32P; else s->avctx->sample_fmt = AV_SAMPLE_FMT_S32; - s->sample_shift = 32 - s->bps; + s->sample_shift = 32 - s->flac_stream_info.bps; } else { if (planar) s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P; else s->avctx->sample_fmt = AV_SAMPLE_FMT_S16; - s->sample_shift = 16 - s->bps; + s->sample_shift = 16 - s->flac_stream_info.bps; } } @@ -101,12 +109,13 @@ static av_cold int flac_decode_init(AVCodecContext *avctx) return AVERROR_INVALIDDATA; /* initialize based on the demuxer-supplied streamdata header */ - ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo); + ff_flac_parse_streaminfo(avctx, &s->flac_stream_info, streaminfo); ret = allocate_buffers(s); if (ret < 0) return ret; flac_set_bps(s); - ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps); + ff_flacdsp_init(&s->dsp, avctx->sample_fmt, + s->flac_stream_info.channels, s->flac_stream_info.bps); s->got_streaminfo = 1; return 0; @@ -124,8 +133,12 @@ static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s) static int allocate_buffers(FLACContext *s) { int buf_size; + int ret; + + av_assert0(s->flac_stream_info.max_blocksize); - buf_size = av_samples_get_buffer_size(NULL, s->channels, s->max_blocksize, + buf_size = av_samples_get_buffer_size(NULL, s->flac_stream_info.channels, + s->flac_stream_info.max_blocksize, AV_SAMPLE_FMT_S32P, 0); if (buf_size < 0) return buf_size; @@ -134,9 +147,12 @@ static int allocate_buffers(FLACContext *s) if (!s->decoded_buffer) return AVERROR(ENOMEM); - return av_samples_fill_arrays((uint8_t **)s->decoded, NULL, - s->decoded_buffer, s->channels, - s->max_blocksize, AV_SAMPLE_FMT_S32P, 0); + ret = av_samples_fill_arrays((uint8_t **)s->decoded, NULL, + s->decoded_buffer, + s->flac_stream_info.channels, + s->flac_stream_info.max_blocksize, + AV_SAMPLE_FMT_S32P, 0); + return ret < 0 ? ret : 0; } /** @@ -159,12 +175,13 @@ static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size) metadata_size != FLAC_STREAMINFO_SIZE) { return AVERROR_INVALIDDATA; } - ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]); + ff_flac_parse_streaminfo(s->avctx, &s->flac_stream_info, &buf[8]); ret = allocate_buffers(s); if (ret < 0) return ret; flac_set_bps(s); - ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps); + ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, + s->flac_stream_info.channels, s->flac_stream_info.bps); s->got_streaminfo = 1; return 0; @@ -213,6 +230,12 @@ static int decode_residuals(FLACContext *s, int32_t *decoded, int pred_order) rice_order = get_bits(&s->gb, 4); samples= s->blocksize >> rice_order; + if (samples << rice_order != s->blocksize) { + av_log(s->avctx, AV_LOG_ERROR, "invalid rice order: %i blocksize %i\n", + rice_order, s->blocksize); + return AVERROR_INVALIDDATA; + } + if (pred_order > samples) { av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n", pred_order, samples); @@ -245,7 +268,8 @@ static int decode_subframe_fixed(FLACContext *s, int32_t *decoded, int pred_order, int bps) { const int blocksize = s->blocksize; - int a, b, c, d, i, ret; + int av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i; + int ret; /* warm up samples */ for (i = 0; i < pred_order; i++) { @@ -291,6 +315,33 @@ static int decode_subframe_fixed(FLACContext *s, int32_t *decoded, return 0; } +static void lpc_analyze_remodulate(int32_t *decoded, const int coeffs[32], + int order, int qlevel, int len, int bps) +{ + int i, j; + int ebps = 1 << (bps-1); + unsigned sigma = 0; + + for (i = order; i < len; i++) + sigma |= decoded[i] + ebps; + + if (sigma < 2*ebps) + return; + + for (i = len - 1; i >= order; i--) { + int64_t p = 0; + for (j = 0; j < order; j++) + p += coeffs[j] * (int64_t)decoded[i-order+j]; + decoded[i] -= p >> qlevel; + } + for (i = order; i < len; i++, decoded++) { + int32_t p = 0; + for (j = 0; j < order; j++) + p += coeffs[j] * (uint32_t)decoded[j]; + decoded[j] += p >> qlevel; + } +} + static int decode_subframe_lpc(FLACContext *s, int32_t *decoded, int pred_order, int bps) { @@ -322,7 +373,15 @@ static int decode_subframe_lpc(FLACContext *s, int32_t *decoded, int pred_order, if ((ret = decode_residuals(s, decoded, pred_order)) < 0) return ret; - s->dsp.lpc(decoded, coeffs, pred_order, qlevel, s->blocksize); + if ( ( s->buggy_lpc && s->flac_stream_info.bps <= 16) + || ( !s->buggy_lpc && bps <= 16 + && bps + coeff_prec + av_log2(pred_order) <= 32)) { + s->dsp.lpc16(decoded, coeffs, pred_order, qlevel, s->blocksize); + } else { + s->dsp.lpc32(decoded, coeffs, pred_order, qlevel, s->blocksize); + if (s->flac_stream_info.bps <= 16) + lpc_analyze_remodulate(decoded, coeffs, pred_order, qlevel, s->blocksize, bps); + } return 0; } @@ -331,7 +390,7 @@ static inline int decode_subframe(FLACContext *s, int channel) { int32_t *decoded = s->decoded[channel]; int type, wasted = 0; - int bps = s->bps; + int bps = s->flac_stream_info.bps; int i, tmp, ret; if (channel == 0) { @@ -350,8 +409,7 @@ static inline int decode_subframe(FLACContext *s, int channel) if (get_bits1(&s->gb)) { int left = get_bits_left(&s->gb); - wasted = 1; - if ( left < 0 || + if ( left <= 0 || (left < bps && !show_bits_long(&s->gb, left)) || !show_bits_long(&s->gb, bps)) { av_log(s->avctx, AV_LOG_ERROR, @@ -359,8 +417,7 @@ static inline int decode_subframe(FLACContext *s, int channel) bps, left); return AVERROR_INVALIDDATA; } - while (!get_bits1(&s->gb)) - wasted++; + wasted = 1 + get_unary(&s->gb, 1, get_bits_left(&s->gb)); bps -= wasted; } if (bps > 32) { @@ -407,66 +464,69 @@ static int decode_frame(FLACContext *s) return ret; } - if (s->channels && fi.channels != s->channels && s->got_streaminfo) { - s->channels = s->avctx->channels = fi.channels; + if ( s->flac_stream_info.channels + && fi.channels != s->flac_stream_info.channels + && s->got_streaminfo) { + s->flac_stream_info.channels = s->avctx->channels = fi.channels; ff_flac_set_channel_layout(s->avctx); ret = allocate_buffers(s); if (ret < 0) return ret; } - s->channels = s->avctx->channels = fi.channels; + s->flac_stream_info.channels = s->avctx->channels = fi.channels; if (!s->avctx->channel_layout) ff_flac_set_channel_layout(s->avctx); s->ch_mode = fi.ch_mode; - if (!s->bps && !fi.bps) { + if (!s->flac_stream_info.bps && !fi.bps) { av_log(s->avctx, AV_LOG_ERROR, "bps not found in STREAMINFO or frame header\n"); return AVERROR_INVALIDDATA; } if (!fi.bps) { - fi.bps = s->bps; - } else if (s->bps && fi.bps != s->bps) { + fi.bps = s->flac_stream_info.bps; + } else if (s->flac_stream_info.bps && fi.bps != s->flac_stream_info.bps) { av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not " "supported\n"); return AVERROR_INVALIDDATA; } - if (!s->bps) { - s->bps = s->avctx->bits_per_raw_sample = fi.bps; + if (!s->flac_stream_info.bps) { + s->flac_stream_info.bps = s->avctx->bits_per_raw_sample = fi.bps; flac_set_bps(s); } - if (!s->max_blocksize) - s->max_blocksize = FLAC_MAX_BLOCKSIZE; - if (fi.blocksize > s->max_blocksize) { + if (!s->flac_stream_info.max_blocksize) + s->flac_stream_info.max_blocksize = FLAC_MAX_BLOCKSIZE; + if (fi.blocksize > s->flac_stream_info.max_blocksize) { av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", fi.blocksize, - s->max_blocksize); + s->flac_stream_info.max_blocksize); return AVERROR_INVALIDDATA; } s->blocksize = fi.blocksize; - if (!s->samplerate && !fi.samplerate) { + if (!s->flac_stream_info.samplerate && !fi.samplerate) { av_log(s->avctx, AV_LOG_ERROR, "sample rate not found in STREAMINFO" " or frame header\n"); return AVERROR_INVALIDDATA; } if (fi.samplerate == 0) - fi.samplerate = s->samplerate; - s->samplerate = s->avctx->sample_rate = fi.samplerate; + fi.samplerate = s->flac_stream_info.samplerate; + s->flac_stream_info.samplerate = s->avctx->sample_rate = fi.samplerate; if (!s->got_streaminfo) { ret = allocate_buffers(s); if (ret < 0) return ret; - ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps); s->got_streaminfo = 1; - dump_headers(s->avctx, (FLACStreaminfo *)s); + dump_headers(s->avctx, &s->flac_stream_info); } + ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, + s->flac_stream_info.channels, s->flac_stream_info.bps); -// dump_headers(s->avctx, (FLACStreaminfo *)s); +// dump_headers(s->avctx, &s->flac_stream_info); /* subframes */ - for (i = 0; i < s->channels; i++) { + for (i = 0; i < s->flac_stream_info.channels; i++) { if ((ret = decode_subframe(s, i)) < 0) return ret; } @@ -483,6 +543,7 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { AVFrame *frame = data; + ThreadFrame tframe = { .f = data }; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; FLACContext *s = avctx->priv_data; @@ -491,12 +552,22 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data, *got_frame_ptr = 0; - if (s->max_framesize == 0) { - s->max_framesize = - ff_flac_get_max_frame_size(s->max_blocksize ? s->max_blocksize : FLAC_MAX_BLOCKSIZE, + if (s->flac_stream_info.max_framesize == 0) { + s->flac_stream_info.max_framesize = + ff_flac_get_max_frame_size(s->flac_stream_info.max_blocksize ? s->flac_stream_info.max_blocksize : FLAC_MAX_BLOCKSIZE, FLAC_MAX_CHANNELS, 32); } + if (buf_size > 5 && !memcmp(buf, "\177FLAC", 5)) { + av_log(s->avctx, AV_LOG_DEBUG, "skipping flac header packet 1\n"); + return buf_size; + } + + if (buf_size > 0 && (*buf & 0x7F) == FLAC_METADATA_TYPE_VORBIS_COMMENT) { + av_log(s->avctx, AV_LOG_DEBUG, "skipping vorbis comment\n"); + return buf_size; + } + /* check that there is at least the smallest decodable amount of data. this amount corresponds to the smallest valid FLAC frame possible. FF F8 69 02 00 00 9A 00 00 34 46 */ @@ -513,21 +584,29 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data, } /* decode frame */ - init_get_bits(&s->gb, buf, buf_size*8); + if ((ret = init_get_bits8(&s->gb, buf, buf_size)) < 0) + return ret; if ((ret = decode_frame(s)) < 0) { av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n"); return ret; } - bytes_read = (get_bits_count(&s->gb)+7)/8; + bytes_read = get_bits_count(&s->gb)/8; + + if ((s->avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_COMPLIANT)) && + av_crc(av_crc_get_table(AV_CRC_16_ANSI), + 0, buf, bytes_read)) { + av_log(s->avctx, AV_LOG_ERROR, "CRC error at PTS %"PRId64"\n", avpkt->pts); + if (s->avctx->err_recognition & AV_EF_EXPLODE) + return AVERROR_INVALIDDATA; + } /* get output buffer */ frame->nb_samples = s->blocksize; - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { - av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + if ((ret = ff_thread_get_buffer(avctx, &tframe, 0)) < 0) return ret; - } - s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded, s->channels, + s->dsp.decorrelate[s->ch_mode](frame->data, s->decoded, + s->flac_stream_info.channels, s->blocksize, s->sample_shift); if (bytes_read > buf_size) { @@ -544,6 +623,17 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data, return bytes_read; } +static int init_thread_copy(AVCodecContext *avctx) +{ + FLACContext *s = avctx->priv_data; + s->decoded_buffer = NULL; + s->decoded_buffer_size = 0; + s->avctx = avctx; + if (s->flac_stream_info.max_blocksize) + return allocate_buffers(s); + return 0; +} + static av_cold int flac_decode_close(AVCodecContext *avctx) { FLACContext *s = avctx->priv_data; @@ -553,6 +643,18 @@ static av_cold int flac_decode_close(AVCodecContext *avctx) return 0; } +static const AVOption options[] = { +{ "use_buggy_lpc", "emulate old buggy lavc behavior", offsetof(FLACContext, buggy_lpc), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, +{ NULL }, +}; + +static const AVClass flac_decoder_class = { + "FLAC decoder", + av_default_item_name, + options, + LIBAVUTIL_VERSION_INT, +}; + AVCodec ff_flac_decoder = { .name = "flac", .long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"), @@ -562,10 +664,12 @@ AVCodec ff_flac_decoder = { .init = flac_decode_init, .close = flac_decode_close, .decode = flac_decode_frame, - .capabilities = CODEC_CAP_DR1, + .init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy), + .capabilities = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P, - -1 }, + AV_SAMPLE_FMT_NONE }, + .priv_class = &flac_decoder_class, }; |
