/* * Shorten decoder * Copyright (c) 2005 Jeff Muizelaar * * 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 * Shorten decoder * @author Jeff Muizelaar * */ #include <limits.h> #include "avcodec.h" #include "bytestream.h" #include "get_bits.h" #include "golomb.h" #include "internal.h" #define MAX_CHANNELS 8 #define MAX_BLOCKSIZE 65535 #define OUT_BUFFER_SIZE 16384 #define ULONGSIZE 2 #define WAVE_FORMAT_PCM 0x0001 #define DEFAULT_BLOCK_SIZE 256 #define TYPESIZE 4 #define CHANSIZE 0 #define LPCQSIZE 2 #define ENERGYSIZE 3 #define BITSHIFTSIZE 2 #define TYPE_S8 1 #define TYPE_U8 2 #define TYPE_S16HL 3 #define TYPE_U16HL 4 #define TYPE_S16LH 5 #define TYPE_U16LH 6 #define NWRAP 3 #define NSKIPSIZE 1 #define LPCQUANT 5 #define V2LPCQOFFSET (1 << LPCQUANT) #define FNSIZE 2 #define FN_DIFF0 0 #define FN_DIFF1 1 #define FN_DIFF2 2 #define FN_DIFF3 3 #define FN_QUIT 4 #define FN_BLOCKSIZE 5 #define FN_BITSHIFT 6 #define FN_QLPC 7 #define FN_ZERO 8 #define FN_VERBATIM 9 /** indicates if the FN_* command is audio or non-audio */ static const uint8_t is_audio_command[10] = { 1, 1, 1, 1, 0, 0, 0, 1, 1, 0 }; #define VERBATIM_CKSIZE_SIZE 5 #define VERBATIM_BYTE_SIZE 8 #define CANONICAL_HEADER_SIZE 44 typedef struct ShortenContext { AVCodecContext *avctx; GetBitContext gb; int min_framesize, max_framesize; unsigned channels; int32_t *decoded[MAX_CHANNELS]; int32_t *decoded_base[MAX_CHANNELS]; int32_t *offset[MAX_CHANNELS]; int *coeffs; uint8_t *bitstream; int bitstream_size; int bitstream_index; unsigned int allocated_bitstream_size; int header_size; uint8_t header[OUT_BUFFER_SIZE]; int version; int cur_chan; int bitshift; int nmean; int internal_ftype; int nwrap; int blocksize; int bitindex; int32_t lpcqoffset; int got_header; int got_quit_command; } ShortenContext; static av_cold int shorten_decode_init(AVCodecContext *avctx) { ShortenContext *s = avctx->priv_data; s->avctx = avctx; return 0; } static int allocate_buffers(ShortenContext *s) { int i, chan; int *coeffs; void *tmp_ptr; for (chan = 0; chan < s->channels; chan++) { if (FFMAX(1, s->nmean) >= UINT_MAX / sizeof(int32_t)) { av_log(s->avctx, AV_LOG_ERROR, "nmean too large\n"); return AVERROR_INVALIDDATA; } if (s->blocksize + s->nwrap >= UINT_MAX / sizeof(int32_t) || s->blocksize + s->nwrap <= (unsigned)s->nwrap) { av_log(s->avctx, AV_LOG_ERROR, "s->blocksize + s->nwrap too large\n"); return AVERROR_INVALIDDATA; } tmp_ptr = av_realloc(s->offset[chan], sizeof(int32_t) * FFMAX(1, s->nmean)); if (!tmp_ptr) return AVERROR(ENOMEM); s->offset[chan] = tmp_ptr; tmp_ptr = av_realloc(s->decoded_base[chan], (s->blocksize + s->nwrap) * sizeof(s->decoded_base[0][0])); if (!tmp_ptr) return AVERROR(ENOMEM); s->decoded_base[chan] = tmp_ptr; for (i = 0; i < s->nwrap; i++) s->decoded_base[chan][i] = 0; s->decoded[chan] = s->decoded_base[chan] + s->nwrap; } coeffs = av_realloc(s->coeffs, s->nwrap * sizeof(*s->coeffs)); if (!coeffs) return AVERROR(ENOMEM); s->coeffs = coeffs; return 0; } static inline unsigned int get_uint(ShortenContext *s, int k) { if (s->version != 0) k = get_ur_golomb_shorten(&s->gb, ULONGSIZE); return get_ur_golomb_shorten(&s->gb, k); } static void fix_bitshift(ShortenContext *s, int32_t *buffer) { int i; if (s->bitshift != 0) for (i = 0; i < s->blocksize; i++) buffer[i] <<= s->bitshift; } static int init_offset(ShortenContext *s) { int32_t mean = 0; int chan, i; int nblock = FFMAX(1, s->nmean); /* initialise offset */ switch (s->internal_ftype) { case TYPE_U8: s->avctx->sample_fmt = AV_SAMPLE_FMT_U8P; mean = 0x80; break; case TYPE_S16HL: case TYPE_S16LH: s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P; break; default: av_log(s->avctx, AV_LOG_ERROR, "unknown audio type\n"); return AVERROR_PATCHWELCOME; } for (chan = 0; chan < s->channels; chan++) for (i = 0; i < nblock; i++) s->offset[chan][i] = mean; return 0; } static int decode_wave_header(AVCodecContext *avctx, const uint8_t *header, int header_size) { int len, bps; short wave_format; GetByteContext gb; bytestream2_init(&gb, header, header_size); if (bytestream2_get_le32(&gb) != MKTAG('R', 'I', 'F', 'F')) { av_log(avctx, AV_LOG_ERROR, "missing RIFF tag\n"); return AVERROR_INVALIDDATA; } bytestream2_skip(&gb, 4); /* chunk size */ if (bytestream2_get_le32(&gb) != MKTAG('W', 'A', 'V', 'E')) { av_log(avctx, AV_LOG_ERROR, "missing WAVE tag\n"); return AVERROR_INVALIDDATA; } while (bytestream2_get_le32(&gb) != MKTAG('f', 'm', 't', ' ')) { len = bytestream2_get_le32(&gb); bytestream2_skip(&gb, len); if (len < 0 || bytestream2_get_bytes_left(&gb) < 16) { av_log(avctx, AV_LOG_ERROR, "no fmt chunk found\n"); return AVERROR_INVALIDDATA; } } len = bytestream2_get_le32(&gb); if (len < 16) { av_log(avctx, AV_LOG_ERROR, "fmt chunk was too short\n"); return AVERROR_INVALIDDATA; } wave_format = bytestream2_get_le16(&gb); switch (wave_format) { case WAVE_FORMAT_PCM: break; default: av_log(avctx, AV_LOG_ERROR, "unsupported wave format\n"); return AVERROR(ENOSYS); } bytestream2_skip(&gb, 2); // skip channels (already got from shorten header) avctx->sample_rate = bytestream2_get_le32(&gb); bytestream2_skip(&gb, 4); // skip bit rate (represents original uncompressed bit rate) bytestream2_skip(&gb, 2); // skip block align (not needed) bps = bytestream2_get_le16(&gb); avctx->bits_per_coded_sample = bps; if (bps != 16 && bps != 8) { av_log(avctx, AV_LOG_ERROR, "unsupported number of bits per sample: %d\n", bps); return AVERROR(ENOSYS); } len -= 16; if (len > 0) av_log(avctx, AV_LOG_INFO, "%d header bytes unparsed\n", len); return 0; } static const int fixed_coeffs[][3] = { { 0, 0, 0 }, { 1, 0, 0 }, { 2, -1, 0 }, { 3, -3, 1 } }; static int decode_subframe_lpc(ShortenContext *s, int command, int channel, int residual_size, int32_t coffset) { int pred_order, sum, qshift, init_sum, i, j; const int *coeffs; if (command == FN_QLPC) { /* read/validate prediction order */ pred_order = get_ur_golomb_shorten(&s->gb, LPCQSIZE); if (pred_order > s->nwrap) { av_log(s->avctx, AV_LOG_ERROR, "invalid pred_order %d\n", pred_order); return AVERROR(EINVAL); } /* read LPC coefficients */ for (i = 0; i < pred_order; i++) s->coeffs[i] = get_sr_golomb_shorten(&s->gb, LPCQUANT); coeffs = s->coeffs; qshift = LPCQUANT; } else { /* fixed LPC coeffs */ pred_order = command; if (pred_order >= FF_ARRAY_ELEMS(fixed_coeffs)) { av_log(s->avctx, AV_LOG_ERROR, "invalid pred_order %d\n", pred_order); return AVERROR_INVALIDDATA; } coeffs = fixed_coeffs[pred_order]; qshift = 0; } /* subtract offset from previous samples to use in prediction */ if (command == FN_QLPC && coffset) for (i = -pred_order; i < 0; i++) s->decoded[channel][i] -= coffset; /* decode residual and do LPC prediction */ init_sum = pred_order ? (command == FN_QLPC ? s->lpcqoffset : 0) : coffset; for (i = 0; i < s->blocksize; i++) { sum = init_sum; for (j = 0; j < pred_order; j++) sum += coeffs[j] * s->decoded[channel][i - j - 1]; s->decoded[channel][i] = get_sr_golomb_shorten(&s->gb, residual_size) + (sum >> qshift); } /* add offset to current samples */ if (command == FN_QLPC && coffset) for (i = 0; i < s->blocksize; i++) s->decoded[channel][i] += coffset; return 0; } static int read_header(ShortenContext *s) { int i, ret; int maxnlpc = 0; /* shorten signature */ if (get_bits_long(&s->gb, 32) != AV_RB32("ajkg")) { av_log(s->avctx, AV_LOG_ERROR, "missing shorten magic 'ajkg'\n"); return AVERROR_INVALIDDATA; } s->lpcqoffset = 0; s->blocksize = DEFAULT_BLOCK_SIZE; s->nmean = -1; s->version = get_bits(&s->gb, 8); s->internal_ftype = get_uint(s, TYPESIZE); s->channels = get_uint(s, CHANSIZE); if (!s->channels) { av_log(s->avctx, AV_LOG_ERROR, "No channels reported\n"); return AVERROR_INVALIDDATA; } if (s->channels > MAX_CHANNELS) { av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels); s->channels = 0; return AVERROR_INVALIDDATA; } s->avctx->channels = s->channels; /* get blocksize if version > 0 */ if (s->version > 0) { int skip_bytes; unsigned blocksize; blocksize = get_uint(s, av_log2(DEFAULT_BLOCK_SIZE)); if (!blocksize || blocksize > MAX_BLOCKSIZE) { av_log(s->avctx, AV_LOG_ERROR, "invalid or unsupported block size: %d\n", blocksize); return AVERROR(EINVAL); } s->blocksize = blocksize; maxnlpc = get_uint(s, LPCQSIZE); s->nmean = get_uint(s, 0); skip_bytes = get_uint(s, NSKIPSIZE); for (i = 0; i < skip_bytes; i++) skip_bits(&s->gb, 8); } s->nwrap = FFMAX(NWRAP, maxnlpc); if ((ret = allocate_buffers(s)) < 0) return ret; if ((ret = init_offset(s)) < 0) return ret; if (s->version > 1) s->lpcqoffset = V2LPCQOFFSET; if (get_ur_golomb_shorten(&s->gb, FNSIZE) != FN_VERBATIM) { av_log(s->avctx, AV_LOG_ERROR, "missing verbatim section at beginning of stream\n"); return AVERROR_INVALIDDATA; } s->header_size = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE); if (s->header_size >= OUT_BUFFER_SIZE || s->header_size < CANONICAL_HEADER_SIZE) { av_log(s->avctx, AV_LOG_ERROR, "header is wrong size: %d\n", s->header_size); return AVERROR_INVALIDDATA; } for (i = 0; i < s->header_size; i++) s->header[i] = (char)get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE); if ((ret = decode_wave_header(s->avctx, s->header, s->header_size)) < 0) return ret; s->cur_chan = 0; s->bitshift = 0; s->got_header = 1; return 0; } static int shorten_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { AVFrame *frame = data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; ShortenContext *s = avctx->priv_data; int i, input_buf_size = 0; int ret; /* allocate internal bitstream buffer */ if (s->max_framesize == 0) { void *tmp_ptr; s->max_framesize = 8192; // should hopefully be enough for the first header tmp_ptr = av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize + FF_INPUT_BUFFER_PADDING_SIZE); if (!tmp_ptr) { av_log(avctx, AV_LOG_ERROR, "error allocating bitstream buffer\n"); return AVERROR(ENOMEM); } s->bitstream = tmp_ptr; } /* append current packet data to bitstream buffer */ if (1 && s->max_framesize) { //FIXME truncated buf_size = FFMIN(buf_size, s->max_framesize - s->bitstream_size); input_buf_size = buf_size; if (s->bitstream_index + s->bitstream_size + buf_size + FF_INPUT_BUFFER_PADDING_SIZE > s->allocated_bitstream_size) { memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size); s->bitstream_index = 0; } if (buf) memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size); buf = &s->bitstream[s->bitstream_index]; buf_size += s->bitstream_size; s->bitstream_size = buf_size; /* do not decode until buffer has at least max_framesize bytes or * the end of the file has been reached */ if (buf_size < s->max_framesize && avpkt->data) { *got_frame_ptr = 0; return input_buf_size; } } /* init and position bitstream reader */ init_get_bits(&s->gb, buf, buf_size * 8); skip_bits(&s->gb, s->bitindex); /* process header or next subblock */ if (!s->got_header) { if ((ret = read_header(s)) < 0) return ret; *got_frame_ptr = 0; goto finish_frame; } /* if quit command was read previously, don't decode anything */ if (s->got_quit_command) { *got_frame_ptr = 0; return avpkt->size; } s->cur_chan = 0; while (s->cur_chan < s->channels) { unsigned cmd; int len; if (get_bits_left(&s->gb) < 3 + FNSIZE) { *got_frame_ptr = 0; break; } cmd = get_ur_golomb_shorten(&s->gb, FNSIZE); if (cmd > FN_VERBATIM) { av_log(avctx, AV_LOG_ERROR, "unknown shorten function %d\n", cmd); *got_frame_ptr = 0; break; } if (!is_audio_command[cmd]) { /* process non-audio command */ switch (cmd) { case FN_VERBATIM: len = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE); while (len--) get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE); break; case FN_BITSHIFT: s->bitshift = get_ur_golomb_shorten(&s->gb, BITSHIFTSIZE); break; case FN_BLOCKSIZE: { unsigned blocksize = get_uint(s, av_log2(s->blocksize)); if (blocksize > s->blocksize) { av_log(avctx, AV_LOG_ERROR, "Increasing block size is not supported\n"); return AVERROR_PATCHWELCOME; } if (!blocksize || blocksize > MAX_BLOCKSIZE) { av_log(avctx, AV_LOG_ERROR, "invalid or unsupported " "block size: %d\n", blocksize); return AVERROR(EINVAL); } s->blocksize = blocksize; break; } case FN_QUIT: s->got_quit_command = 1; break; } if (cmd == FN_BLOCKSIZE || cmd == FN_QUIT) { *got_frame_ptr = 0; break; } } else { /* process audio command */ int residual_size = 0; int channel = s->cur_chan; int32_t coffset; /* get Rice code for residual decoding */ if (cmd != FN_ZERO) { residual_size = get_ur_golomb_shorten(&s->gb, ENERGYSIZE); /* This is a hack as version 0 differed in the definition * of get_sr_golomb_shorten(). */ if (s->version == 0) residual_size--; } /* calculate sample offset using means from previous blocks */ if (s->nmean == 0) coffset = s->offset[channel][0]; else { int32_t sum = (s->version < 2) ? 0 : s->nmean / 2; for (i = 0; i < s->nmean; i++) sum += s->offset[channel][i]; coffset = sum / s->nmean; if (s->version >= 2) coffset = s->bitshift == 0 ? coffset : coffset >> s->bitshift - 1 >> 1; } /* decode samples for this channel */ if (cmd == FN_ZERO) { for (i = 0; i < s->blocksize; i++) s->decoded[channel][i] = 0; } else { if ((ret = decode_subframe_lpc(s, cmd, channel, residual_size, coffset)) < 0) return ret; } /* update means with info from the current block */ if (s->nmean > 0) { int32_t sum = (s->version < 2) ? 0 : s->blocksize / 2; for (i = 0; i < s->blocksize; i++) sum += s->decoded[channel][i]; for (i = 1; i < s->nmean; i++) s->offset[channel][i - 1] = s->offset[channel][i]; if (s->version < 2) s->offset[channel][s->nmean - 1] = sum / s->blocksize; else s->offset[channel][s->nmean - 1] = (sum / s->blocksize) << s->bitshift; } /* copy wrap samples for use with next block */ for (i = -s->nwrap; i < 0; i++) s->decoded[channel][i] = s->decoded[channel][i + s->blocksize]; /* shift samples to add in unused zero bits which were removed * during encoding */ fix_bitshift(s, s->decoded[channel]); /* if this is the last channel in the block, output the samples */ s->cur_chan++; if (s->cur_chan == s->channels) { uint8_t *samples_u8; int16_t *samples_s16; int chan; /* get output buffer */ frame->nb_samples = s->blocksize; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; for (chan = 0; chan < s->channels; chan++) { samples_u8 = ((uint8_t **)frame->extended_data)[chan]; samples_s16 = ((int16_t **)frame->extended_data)[chan]; for (i = 0; i < s->blocksize; i++) { switch (s->internal_ftype) { case TYPE_U8: *samples_u8++ = av_clip_uint8(s->decoded[chan][i]); break; case TYPE_S16HL: case TYPE_S16LH: *samples_s16++ = av_clip_int16(s->decoded[chan][i]); break; } } } *got_frame_ptr = 1; } } } if (s->cur_chan < s->channels) *got_frame_ptr = 0; finish_frame: s->bitindex = get_bits_count(&s->gb) - 8 * (get_bits_count(&s->gb) / 8); i = get_bits_count(&s->gb) / 8; if (i > buf_size) { av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size); s->bitstream_size = 0; s->bitstream_index = 0; return AVERROR_INVALIDDATA; } if (s->bitstream_size) { s->bitstream_index += i; s->bitstream_size -= i; return input_buf_size; } else return i; } static av_cold int shorten_decode_close(AVCodecContext *avctx) { ShortenContext *s = avctx->priv_data; int i; for (i = 0; i < s->channels; i++) { s->decoded[i] = NULL; av_freep(&s->decoded_base[i]); av_freep(&s->offset[i]); } av_freep(&s->bitstream); av_freep(&s->coeffs); return 0; } AVCodec ff_shorten_decoder = { .name = "shorten", .long_name = NULL_IF_CONFIG_SMALL("Shorten"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_SHORTEN, .priv_data_size = sizeof(ShortenContext), .init = shorten_decode_init, .close = shorten_decode_close, .decode = shorten_decode_frame, .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_NONE }, };