diff options
Diffstat (limited to 'media/ffvpx/libavcodec/flacdec.c')
| -rw-r--r-- | media/ffvpx/libavcodec/flacdec.c | 677 |
1 files changed, 677 insertions, 0 deletions
diff --git a/media/ffvpx/libavcodec/flacdec.c b/media/ffvpx/libavcodec/flacdec.c new file mode 100644 index 000000000..b7237e18f --- /dev/null +++ b/media/ffvpx/libavcodec/flacdec.c @@ -0,0 +1,677 @@ +/* + * FLAC (Free Lossless Audio Codec) decoder + * Copyright (c) 2003 Alex Beregszaszi + * + * This file is part of FFmpeg. + * + * 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. + * + * 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * FLAC (Free Lossless Audio Codec) decoder + * @author Alex Beregszaszi + * @see http://flac.sourceforge.net/ + * + * This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed + * through, starting from the initial 'fLaC' signature; or by passing the + * 34-byte streaminfo structure through avctx->extradata[_size] followed + * by data starting with the 0xFFF8 marker. + */ + +#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" +#include "bytestream.h" +#include "golomb.h" +#include "flac.h" +#include "flacdata.h" +#include "flacdsp.h" +#include "thread.h" +#include "unary.h" + + +typedef struct FLACContext { + AVClass *class; + struct FLACStreaminfo flac_stream_info; + + AVCodecContext *avctx; ///< parent AVCodecContext + GetBitContext gb; ///< GetBitContext initialized to start at the current frame + + int blocksize; ///< number of samples in the current frame + int sample_shift; ///< shift required to make output samples 16-bit or 32-bit + int ch_mode; ///< channel decorrelation type in the current frame + int got_streaminfo; ///< indicates if the STREAMINFO has been read + + 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; + +static int allocate_buffers(FLACContext *s); + +static void flac_set_bps(FLACContext *s) +{ + enum AVSampleFormat req = s->avctx->request_sample_fmt; + 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); + + if (need32 || want32) { + if (planar) + s->avctx->sample_fmt = AV_SAMPLE_FMT_S32P; + else + s->avctx->sample_fmt = AV_SAMPLE_FMT_S32; + 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->flac_stream_info.bps; + } +} + +static av_cold int flac_decode_init(AVCodecContext *avctx) +{ + enum FLACExtradataFormat format; + uint8_t *streaminfo; + int ret; + FLACContext *s = avctx->priv_data; + s->avctx = avctx; + + /* for now, the raw FLAC header is allowed to be passed to the decoder as + frame data instead of extradata. */ + if (!avctx->extradata) + return 0; + + if (!ff_flac_is_extradata_valid(avctx, &format, &streaminfo)) + return AVERROR_INVALIDDATA; + + /* initialize based on the demuxer-supplied streamdata header */ + 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->flac_stream_info.channels, s->flac_stream_info.bps); + s->got_streaminfo = 1; + + return 0; +} + +static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s) +{ + av_log(avctx, AV_LOG_DEBUG, " Max Blocksize: %d\n", s->max_blocksize); + av_log(avctx, AV_LOG_DEBUG, " Max Framesize: %d\n", s->max_framesize); + av_log(avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate); + av_log(avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels); + av_log(avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps); +} + +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->flac_stream_info.channels, + s->flac_stream_info.max_blocksize, + AV_SAMPLE_FMT_S32P, 0); + if (buf_size < 0) + return buf_size; + + av_fast_malloc(&s->decoded_buffer, &s->decoded_buffer_size, buf_size); + if (!s->decoded_buffer) + return AVERROR(ENOMEM); + + 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; +} + +/** + * Parse the STREAMINFO from an inline header. + * @param s the flac decoding context + * @param buf input buffer, starting with the "fLaC" marker + * @param buf_size buffer size + * @return non-zero if metadata is invalid + */ +static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size) +{ + int metadata_type, metadata_size, ret; + + if (buf_size < FLAC_STREAMINFO_SIZE+8) { + /* need more data */ + return 0; + } + flac_parse_block_header(&buf[4], NULL, &metadata_type, &metadata_size); + if (metadata_type != FLAC_METADATA_TYPE_STREAMINFO || + metadata_size != FLAC_STREAMINFO_SIZE) { + return AVERROR_INVALIDDATA; + } + 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->flac_stream_info.channels, s->flac_stream_info.bps); + s->got_streaminfo = 1; + + return 0; +} + +/** + * Determine the size of an inline header. + * @param buf input buffer, starting with the "fLaC" marker + * @param buf_size buffer size + * @return number of bytes in the header, or 0 if more data is needed + */ +static int get_metadata_size(const uint8_t *buf, int buf_size) +{ + int metadata_last, metadata_size; + const uint8_t *buf_end = buf + buf_size; + + buf += 4; + do { + if (buf_end - buf < 4) + return 0; + flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size); + buf += 4; + if (buf_end - buf < metadata_size) { + /* need more data in order to read the complete header */ + return 0; + } + buf += metadata_size; + } while (!metadata_last); + + return buf_size - (buf_end - buf); +} + +static int decode_residuals(FLACContext *s, int32_t *decoded, int pred_order) +{ + int i, tmp, partition, method_type, rice_order; + int rice_bits, rice_esc; + int samples; + + method_type = get_bits(&s->gb, 2); + if (method_type > 1) { + av_log(s->avctx, AV_LOG_ERROR, "illegal residual coding method %d\n", + method_type); + return AVERROR_INVALIDDATA; + } + + 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); + return AVERROR_INVALIDDATA; + } + + rice_bits = 4 + method_type; + rice_esc = (1 << rice_bits) - 1; + + decoded += pred_order; + i= pred_order; + for (partition = 0; partition < (1 << rice_order); partition++) { + tmp = get_bits(&s->gb, rice_bits); + if (tmp == rice_esc) { + tmp = get_bits(&s->gb, 5); + for (; i < samples; i++) + *decoded++ = get_sbits_long(&s->gb, tmp); + } else { + for (; i < samples; i++) { + *decoded++ = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0); + } + } + i= 0; + } + + return 0; +} + +static int decode_subframe_fixed(FLACContext *s, int32_t *decoded, + int pred_order, int bps) +{ + const int blocksize = s->blocksize; + 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++) { + decoded[i] = get_sbits_long(&s->gb, bps); + } + + if ((ret = decode_residuals(s, decoded, pred_order)) < 0) + return ret; + + if (pred_order > 0) + a = decoded[pred_order-1]; + if (pred_order > 1) + b = a - decoded[pred_order-2]; + if (pred_order > 2) + c = b - decoded[pred_order-2] + decoded[pred_order-3]; + if (pred_order > 3) + d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4]; + + switch (pred_order) { + case 0: + break; + case 1: + for (i = pred_order; i < blocksize; i++) + decoded[i] = a += decoded[i]; + break; + case 2: + for (i = pred_order; i < blocksize; i++) + decoded[i] = a += b += decoded[i]; + break; + case 3: + for (i = pred_order; i < blocksize; i++) + decoded[i] = a += b += c += decoded[i]; + break; + case 4: + for (i = pred_order; i < blocksize; i++) + decoded[i] = a += b += c += d += decoded[i]; + break; + default: + av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order); + return AVERROR_INVALIDDATA; + } + + 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) +{ + int i, ret; + int coeff_prec, qlevel; + int coeffs[32]; + + /* warm up samples */ + for (i = 0; i < pred_order; i++) { + decoded[i] = get_sbits_long(&s->gb, bps); + } + + coeff_prec = get_bits(&s->gb, 4) + 1; + if (coeff_prec == 16) { + av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n"); + return AVERROR_INVALIDDATA; + } + qlevel = get_sbits(&s->gb, 5); + if (qlevel < 0) { + av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n", + qlevel); + return AVERROR_INVALIDDATA; + } + + for (i = 0; i < pred_order; i++) { + coeffs[pred_order - i - 1] = get_sbits(&s->gb, coeff_prec); + } + + if ((ret = decode_residuals(s, decoded, pred_order)) < 0) + return ret; + + 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; +} + +static inline int decode_subframe(FLACContext *s, int channel) +{ + int32_t *decoded = s->decoded[channel]; + int type, wasted = 0; + int bps = s->flac_stream_info.bps; + int i, tmp, ret; + + if (channel == 0) { + if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE) + bps++; + } else { + if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE) + bps++; + } + + if (get_bits1(&s->gb)) { + av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n"); + return AVERROR_INVALIDDATA; + } + type = get_bits(&s->gb, 6); + + if (get_bits1(&s->gb)) { + int left = get_bits_left(&s->gb); + if ( left <= 0 || + (left < bps && !show_bits_long(&s->gb, left)) || + !show_bits_long(&s->gb, bps)) { + av_log(s->avctx, AV_LOG_ERROR, + "Invalid number of wasted bits > available bits (%d) - left=%d\n", + bps, left); + return AVERROR_INVALIDDATA; + } + wasted = 1 + get_unary(&s->gb, 1, get_bits_left(&s->gb)); + bps -= wasted; + } + if (bps > 32) { + avpriv_report_missing_feature(s->avctx, "Decorrelated bit depth > 32"); + return AVERROR_PATCHWELCOME; + } + +//FIXME use av_log2 for types + if (type == 0) { + tmp = get_sbits_long(&s->gb, bps); + for (i = 0; i < s->blocksize; i++) + decoded[i] = tmp; + } else if (type == 1) { + for (i = 0; i < s->blocksize; i++) + decoded[i] = get_sbits_long(&s->gb, bps); + } else if ((type >= 8) && (type <= 12)) { + if ((ret = decode_subframe_fixed(s, decoded, type & ~0x8, bps)) < 0) + return ret; + } else if (type >= 32) { + if ((ret = decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps)) < 0) + return ret; + } else { + av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n"); + return AVERROR_INVALIDDATA; + } + + if (wasted) { + int i; + for (i = 0; i < s->blocksize; i++) + decoded[i] <<= wasted; + } + + return 0; +} + +static int decode_frame(FLACContext *s) +{ + int i, ret; + GetBitContext *gb = &s->gb; + FLACFrameInfo fi; + + if ((ret = ff_flac_decode_frame_header(s->avctx, gb, &fi, 0)) < 0) { + av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n"); + return ret; + } + + 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->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->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->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->flac_stream_info.bps) { + s->flac_stream_info.bps = s->avctx->bits_per_raw_sample = fi.bps; + flac_set_bps(s); + } + + 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->flac_stream_info.max_blocksize); + return AVERROR_INVALIDDATA; + } + s->blocksize = fi.blocksize; + + 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->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; + s->got_streaminfo = 1; + 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, &s->flac_stream_info); + + /* subframes */ + for (i = 0; i < s->flac_stream_info.channels; i++) { + if ((ret = decode_subframe(s, i)) < 0) + return ret; + } + + align_get_bits(gb); + + /* frame footer */ + skip_bits(gb, 16); /* data crc */ + + return 0; +} + +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; + int bytes_read = 0; + int ret; + + *got_frame_ptr = 0; + + 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 */ + if (buf_size < FLAC_MIN_FRAME_SIZE) + return buf_size; + + /* check for inline header */ + if (AV_RB32(buf) == MKBETAG('f','L','a','C')) { + if (!s->got_streaminfo && (ret = parse_streaminfo(s, buf, buf_size))) { + av_log(s->avctx, AV_LOG_ERROR, "invalid header\n"); + return ret; + } + return get_metadata_size(buf, buf_size); + } + + /* decode frame */ + 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)/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_thread_get_buffer(avctx, &tframe, 0)) < 0) + return ret; + + 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) { + av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size); + return AVERROR_INVALIDDATA; + } + if (bytes_read < buf_size) { + av_log(s->avctx, AV_LOG_DEBUG, "underread: %d orig size: %d\n", + buf_size - bytes_read, buf_size); + } + + *got_frame_ptr = 1; + + return bytes_read; +} + +#if HAVE_THREADS +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; +} +#endif + +static av_cold int flac_decode_close(AVCodecContext *avctx) +{ + FLACContext *s = avctx->priv_data; + + av_freep(&s->decoded_buffer); + + return 0; +} + +static const AVOption options[] = { +{ "use_buggy_lpc", "emulate old buggy lavc behavior", offsetof(FLACContext, buggy_lpc), AV_OPT_TYPE_BOOL, {.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)"), + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_FLAC, + .priv_data_size = sizeof(FLACContext), + .init = flac_decode_init, + .close = flac_decode_close, + .decode = flac_decode_frame, + .init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy), + .capabilities = AV_CODEC_CAP_DR1 | AV_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, + AV_SAMPLE_FMT_NONE }, + .priv_class = &flac_decoder_class, +}; 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